Paper 3 370
Reading ISBN ( for citations)
Telecommunications and Data Communications Handbook
by Ray Horak
John Wiley & Sons © 2007 (832 pages) Citation
ISBN:9780470396070
Chapter 11: Wireless Networking—Emphasis on Mobility
OVERVIEW
The Bell System has available all of the important inventions with reference to commercial wireless telephony. For the usual commercial telephone services in the United States where wires are or can be provided, wireless from both a service and economic standpoint cannot be used. For the supplemental services which cannot be provided by wires, wireless will be used wherever the commercial demand will warrant such use.
Telephone Almanac, American Telephone and Telegraph Company, 1922
In 1876, Alexander Graham Bell demonstrated the telephone at the Centennial Exposition of the United States in Philadelphia, Pennsylvania (United States). From that simple demonstration of one-way transmission over a distance of several hundred feet, the copper-based telephone network grew at an astounding rate. In 1880, Bell invented the first wireless communications system using reflected sunlight and photoelectric selenium receivers. Using this technique, however impractical, he was able to transmit intelligible speech a distance of up to 700 ft. Bell named this invention the photophone, later renaming it the radiophone, which he described as his greatest invention [1, 2]. AT&T continued work on the technology, extending its reach to several miles. The German Nazi military experimented with similar but more advanced systems in tank warfare during World War II [2].
Toward the end of the nineteenth century and not long after Bell's demonstration, a young German scientist named Heinrich Rudolf Hertz discovered the phe-nomenon of invisible force waves emanating for several meters around an electric spark of sufficient intensity. Classical physicists, at a loss to explain the phenomenon, theorized the existence of an unknown medium, luminiferous ether, that conducted that signal. Shortly thereafter, Guglielmo Marconi transmitted these Hertzian waves over several kilometers; he named the new technology radio because the waves appeared to radiate from the transmitter [2]. In 1886, Marconi was granted a patent for the first practical wireless telegraph, for which he shared in the 1909 Nobel Prize in Physics. In the meantime (1900), Canadian scientist Reginald Fessenden (1866–1932) transmitted his own voice a distance of 1 mile over the first wireless telephone [3]. In what he considered to be his greatest achievement in a long and illustrious career, Fessenden successfully broadcast a short program on Christmas Eve, 1906. Having been alerted to a special event, wireless operators of several United Fruit Company ships in the Atlantic heard the first radio audio broadcast in history when Fessenden transmitted Handel's Largo on an Ediphone, played Oh, Holy Night on the violin, and read from the Bible before wishing them a Merry Christmas [4].
The first commercial application of radio technology was that of broadcast radio, introduced in the United States in 1920. The first radio station, the Westinghouse station KDKA in Pittsburgh, Pennsylvania, inaugurated service by broadcasting returns of the Harding–Cox presidential election. On July 25, 1922, the first commercial station, WBAY, which was owned by AT&T, began broadcasting from the AT&T Long Lines building in New York City. Its first paying customer, two months later, was the Queensborough Corporation, advertising its Hawthorne Court real estate development in Jackson Heights. In the meantime, AT&T employees supplied the programming, which consisted of vocal selections, piano recitals, poetry recitations, and other content that seems fairly tame by today's standards. Among the performances was a recitation of James Whitcomb Riley's poem An Old Sweetheart of Mine by Miss Edna Cunningham; there is no record of the audience reaction [1]. While not an immediate commercial success, radio enjoyed U.S. market penetration of 50 percent of households in 1930, 90 percent in 1940, and essentially 100 percent by 1995.
Television, while first demonstrated in 1926 by John Logie Baier, a Scottish inventor, was refined by yet other Bell Labs employees, Frederick Eugene Ives and Frank Gray. The first public demonstration of color TV took place in the United States in 1927, between Bell Labs in Whippany, New Jersey, and Washington, D.C., where the audience included Secretary of Commerce Herbert Hoover. Vaudeville acts were included. Again, there is no record of the reaction of the audience. Delayed by the Great Depression and then World War II, commercial broadcast TV was introduced in 1946 and enjoyed success similar to that of radio [1]. Market penetration in the United States was estimated at 50 percent in 1955, 90 percent in 1960, and 99 percent in 1995. Wireless radio technology also was deployed early on in maritime communications, or ship-to-shore telephony and telegraphy. The first terrestrial mobile application was a one-way system employed in police radio dispatch trials in 1921 at the Detroit, Michigan, Police Department [2, 4].
The technologies and applications discussed in this chapter include Trunk Mobile Radio (TMR), paging, cordless telephony, Wireless Office Telecommunications Systems (WOTS), cellular, Low-Earth Orbiting (LEO) and Middle-Earth Orbiting (MEO) satellites. I discussed Wireless Local Area Networks (WLANs) and Wireless Local Loop (WLL) technologies in Chapters 8 and 9, respectively.
Wireless is about more than being unplugged or untethered. Wireless, in full form, speaks to a fundamentally different way of business and personal communications. Wireless communications adds the element of portability and even mobility, thereby removing the constraints of constant attachments to a physical space such as an office building or residence. Yet, wireless communications technologies generally are relegated to niche applications, at least for the time being. For many applications, wireless just cannot compete with wired networks. Theodore Vail, twice president of AT&T (1880s and early 1900s), summed it up when he said, "The difficulties of the wireless telegraph are as nothing compared with the difficulties in the way of the wireless telephone" [2]. While wireless telephony may have been quite a trick in those days, it is as nothing compared with mobile wireless data communications.
11.1 WIRELESS DEFINED
Wireless, quite simply, refers to communications without wires. While microwave and satellite communications are without wires, those technologies generally are considered to be high-speed network backbone or access technologies that are point to point, point to multipoint, or broadcast in nature. (See Chapter 2 for discussion of the principles and characteristics of radio transmission.) In the context of this discussion, wireless technologies are local loop or local in nature, with the emphasis on mobility.
11.2 STANDARDS AND REGULATIONS
Standards are very important in telecommunications, and wireless is no exception. Wireless technologies have the dubious distinction of lots of standards, many of which are incompatible and conflicting. Existing formal standards are national, regional, and international in nature. There also are proprietary specifications, which sometimes are characterized as de facto standards. Ultimately, you might like to think that the standards wars will yield clear winners and losers, but that is highly unlikely in the immediate future, as standards often are more about politics and economics than technical purity.
Regulation is extremely important in the wireless world, since there is a high potential of interference between transmissions. To avoid this problem, radio must be managed along several dimensions, including frequency allocation and power level. The ITU-R (International Telecommunications Union—Radiocommunication Sector), originally known as the CCIR (Comité consultatif international pour la radio, or International Radio Consultative Committee) is the branch of the ITU that sets international wireless standards. The Institute of Electrical and Electronics Engineers (IEEE) is in the continuing process of developing standards for Wireless LANs through the 802.11 Working Group. The European Telecommunications Standards Institute (ETSI) sets standards within the European Union (EU). The Research for Advanced Communications in Europe (RACE) program, which is directed at the promotion of Integrated Broadband Communications (IBC), addresses wireless in its R1043 recommendation document. The Federal Communications Commission (FCC) has handled regulation of the wireless, as well as the wired, world in the United States since 1934. On a periodic basis, the various national regulatory authorities meet to sort out national and international spectrum allocation issues at the World Radio Conferences (WRC), previously known as the World Administrative Radio Conferences (WARC), which are sponsored by the ITU-R every two years.
Frequency allocation, or spectrum management, involves the designation of certain frequencies in the electromagnetic spectrum in support of certain applications. Examples include AM and FM broadcast radio, UHF and VHF broadcast TV, Trunk Mobile Radio (TMR), cellular radio, and microwave radio. This requirement is essential to avoid interference between various applications using the same, or overlapping, frequency ranges. In limiting each application to a specific range of frequencies, the manufacturers, carriers, and end users of such systems can better be monitored and controlled as well.
The issue of Radio Frequency Interference (RFI), as discussed in Chapter 2, is tied to the specific frequencies employed, the proximity of the transmit/receive antennas, and the power levels involved. As I noted previously, lower frequency signals naturally propagate farther than higher frequency signals, suffering less from attenuation (i.e., loss of signal strength). (Note: This admittedly is a generalization. For example, there are notches in the frequency spectrum that work very poorly due to interaction with water molecules.) In consideration of the specific frequency range employed, therefore, the same frequencies cannot be used by multiple antennas within a certain range of proximity without running the risk of mutual signal interference. The power levels of the various transmitters also must be regulated, as stronger signals propagate farther at any given frequency.
11.3 ADVANTAGES AND DISADVANTAGES OF WIRELESS
Deployment of wireless networks certainly can offer advantages of reduced installation and reconfiguration costs. Tremendous costs can be saved by eliminating requirements to secure terrestrial rights of way, dig trenches and plant poles, place conduits and hang cross-arms, splice cables, place repeaters, and so on. Wireless networks work equally well in rocky or soggy terrain where wired networks may be problematic or even impossible and in remote areas of low user density where wired networks may be impractical. Wireless networks also offer great portability, as the antennas quite easily can be disassembled and reassembled at another location. Wired networks, on the other hand, must be abandoned or removed and sold as scrap. Some wireless networks support mobile applications, with examples being cordless telephony, cellular radio, and packet data radio networks.
Wireless also suffers from certain limitations, perhaps the most significant of which is that of spectrum availability—radio spectrum is a finite resource. The laws of physics and Mother Nature state that radio operates between 30 Hz and 30 GHz (Table 2.1), with the spectrum suitable for communications applications at about 30 kHz and above. While that may seem like a lot of spectrum, there also are a lot of applications and users competing for it, and not all frequency bands are suitable for all applications. This limited radio spectrum divides into even more limited ranges of spectrum allocated in support of specific applications (e.g., microwave and cellular radio). Within each slice of allocated spectrum there clearly exists only so much bandwidth; regardless of how cleverly the engineers and mathematicians design compression algorithms to maximize its use, there remains only so much bandwidth available. Airwave transmission systems also suffer from issues of error performance and security.
There are other problems with wireless communications, by the way. Similar to the effect of echo (i.e., reflected energy) in the wired world, wireless communications suffers from MultiPath Interference (MPI), also known as multipath fading. As radio signals propagate from the transmitter, they naturally diffuse, or spread out, no matter how tightly they are shaped, or focused. If the communication involves a cell phone and a cell site, for example, there really is no shaping. Rather, the signal from the cell site to the terminal is broadcast either on an omnidirectional basis or, more commonly, on a vectored basis. The signal from the terminal back to the cell site always is broadcast on an omnidirectional basis. Assuming that direct line of sight is achievable, some signal elements travel a straight line from transmitter to receiver, while other signal elements bounce off mountains, buildings, cars, trees, your neighbor's dog, and other dense physical objects. This can result in confusion at the receiver. MPI specter refers to the ghosting effect, which occurs when the path of the echo is relatively long and when some reflected signals, therefore, arrive on a significantly delayed basis in a phenomenon known as delay spread. [This ghosting effect is particularly evident in poorly installed coax-based CATV systems, and in traditional antenna-based TV reception in mountainous areas.] To overcome the effects of MPI, the receiver must make comparisons between signals to determine and lock onto the signals of greatest strength, earliest arrival, or both or perhaps to combine the individual weakened signal elements into a single stronger signal.
11.4 CELL CONCEPT: FREQUENCY REUSE
Radio systems are designed for a certain area of coverage, or footprint. Even early radio and TV broadcast systems used the concept to define a service area, such as a metropolitan area. Therefore, you could reuse the same frequencies to support service in metropolitan areas some distance away. For example, 98.1 (MHz) on your FM dial might be WXYZ in New York, New York, and KFRC in San Francisco, California. Similarly, Channel 7 on your TV might be WFAA in Dallas, Texas, and KIRO in Seattle, Washington; broadcast TV stations in the United States can reuse frequencies if separated by at least 150 miles [5]. The size of the cells, of course, is sensitive to frequency and power level as well as the height of the antennas, the topography, the time of day or night, solar activity, weather conditions, and other factors.
Now, I want to digress to my youth. Actually, it is not a complete digression, as it illustrates the points I just made. Some of you may remember Robert ("Bob") Weston Smith. Actually, you may remember him as Wolfman Jack, a famous rock'n roll Disk Jockey (DJ) of the sixties generation and later a successful star in movies such as American Graffiti. (If you don't remember him, just ask your parents.) In any event, as a young man in the 1960s, I would drive hundreds of miles over country roads in Texas to see my girlfriend (which tells you something about the size of the Great State of Texas as well as my difficulty in finding a girlfriend). At 9:00 p.m. on Sunday nights, The Wolfman broadcasted from XERF, the AM radio station in Ciudad Acuna, Coahuila, Mexico, and he came through as "clear as a bell" to the little transistor radio that hung from the rearview mirror of my 1962 Volkswagen beetle. It was not much of a radio (I couldn't afford the kind that fit in the dashboard), but it was a heck of a radio show and a heck of a radio station. XERF, you see, blasted The Wolfman (and a whole host of radio preachers earlier on a Sunday evening) at 250,000 W—five times the allowable power level for radio stations just across the border in Del Rio, Texas, U.S.A. At that power level, given the right weather conditions, The Wolfman could be heard just about as clearly in New York City as I heard him in Corpus Christi. In fact, he overpowered every radio station in North America operating on the same frequency. The Wolfman became famous all over North America, at the expense of the poor DJs who tried to make a living on the same frequency band. Having shared that example with you, I can get back to more serious business. By the way, The Wolfman passed on to that big radio station in the sky on July 1, 1995.
The formal concept of radio cells dates back to 1947, when Bell Telephone engineers developed a radio system concept that included numerous, low-power transmit/receive antennas [1]. Scattered throughout a metropolitan area, this sort of architecture served to increase the effective subscriber capacity of radio systems by breaking the area of coverage into cells, or smaller areas of coverage. Thereby, each frequency could be reused in nonadjacent cells. Additionally, the cells can be split, or subdivided, further as the traffic demands of the system increase. In other words, cellular radio networks are highly scalable.
Frequency reuse is sensitive to factors that are covered in the discussion of microwave and satellite systems in Chapter 2 and that can be seen in the Wolfman Jack example. Specifically, those factors include frequency, power level, antenna design, and topography. Higher frequency signals always attenuate to a greater extent over distance given the same power level. Antenna design is sensitive to wavelength and other factors. Topography is always an issue, as line of sight is always preferable, especially at the higher frequencies in the spectrum. Lower frequencies signals, such as those used in AM radio, tend to follow the curvature of the earth, while higher frequency signals, such as those used in microwave, tend to follow a straight path—straight into space.
11.4.1 Cell Categories
Cells can be characterized as falling into three broad descriptive categories: macro-cells, microcells, and picocells, as illustrated in Figure 11.1. As the cells shrink, the advantages of frequency reuse increase significantly. The increase in traffic-handling capacity can be remarkable, with associated increases in revenue potential. However, the costs of network deployment increase along with the number of base stations and interconnecting links. Also, the complexity of switching mobile traffic between cells increases considerably.
Figure 11.1: Macrocells, microcells, and picocells
Assume that 12 channels are available for use in a metropolitan area that is 60 miles wide (i.e., has a diameter of 60 miles and a radius of 30 miles) and that the terrain is perfectly flat. Assume a seven-cell reuse pattern, which is typical. Consider the following theoretical scenario:
· Macrocells cover relatively large areas, perhaps an entire metropolitan area 60 miles wide. Assume that there is one omnidirectional antenna positioned in the center of the entire metro area and supporting 12 channels. Then only 12 simultaneous conversations can take place within this single macrocell. If the macrocell is divided into seven smaller macrocells in a typical seven-cell reuse pattern, as illustrated in Figures 11.1 and 11.2, with each derived macrocell covering a radius of about 11.3 miles, no frequency reuse is possible and no improvement is realized. Only 12 conversations can be supported due to the fact that the cells must overlap, as illustrated in Figure 11.1. (Note: Radio cells typically are graphically depicted as cells in a honeycomb pattern. Actually, they are irregular, overlapping areas of coverage that are more or less circular or elliptical in form.) Therefore, conversations on the same frequency channels in adjacent cells interfere with each other due to the phenomenon of cochannel interference.
Figure 11.2: Seven-cell reuse pattern
· Microcells cover a smaller area. If each of the derived macrocells were divided into seven equal microcells with a radius of about 4.3 miles, a reuse factor of 7 is realized. In other words, each of the 12 channels could be used seven times and a total of 84 simultaneous conversations could be supported. If the cell size shrinks to 1 mile, the reuse factor is 128, and the same 12 channels could support 1536 simultaneous conversations. (Note: The reuse separation ranges from approximately four to six times the radius of the cells.)
· Picocells are quite small, covering only a few blocks of an urban area or, perhaps, a tunnel, walkway, or parking garage. In a seven-cell reuse pattern, with each cell covering a radius of approximately one-half mile, the reuse factor climbs to 514. In other words, the same 12 channels could theoretically support up to 6168 simultaneous conversations [2].
11.4.2 Cells, Vectors, and Beams
Within a given cell, the antenna may operate in either an omnidirectional or a vectored mode. The omnidirectional approach involves signal transmission from and signal receipt by the centralized base station antenna in all directions. This approach uses all frequency channels in a 360° beam, and all at the same power level. A vectored antenna can subdivide the coverage area of the cell into multiple vectors of coverage. Cellular antennas, for example, commonly carve a cell into three vectors, each of 120°. This vectoring approach enables the carrier frequencies within each vector to be managed independently from those within the other vectors in terms of dimensions such as channel allocation and signal strength.
Smart antennas can improve significantly on this concept of vectoring. Such antennas can operate within a given cell and its constituent vectors, further subdividing the coverage through the use of as many as 12 beams, each with a beam width of 30 °. Thereby, the footprint of the cell can be sculpted to optimize coverage, channel allocation, and interference. The individual beams can rotate from one vector to another and can do so on a dynamic basis in order to adjust to traffic patterns that might vary by time of day, with the process of beam management performed remotely [6].
While I have only a limited amount of space to discuss antenna design, the details of which are the subject for an electrical engineering text, there certainly is at least one other design issue of significance. Code Division Multiple Access (CDMA) systems, which are discussed later in the chapter, deal with issues of multipath interference (MPI) through the use of rake receivers in both the Base Stations (BSs) and Mobile Stations (MSs), for example, cell phones. Rake receivers employ spatial diversity and time diversity in much the same way as Multiple Input, Multiple Output (MIMO) technology in the developing 802.11n WLAN standard discussed in Chapter 8. A rake receiver comprises a set of four receivers, or fingers, that work in a coordinated way to gather signal elements much like the tines of a garden rake work together to gather leaves. Each finger gathers a faded, or attenuated, signal element at a separate moment in time. The receiver combines and correlates the results of all four fingers to optimize the signal, thereby countering the effects of multipath fading and delay spread.
11.5 MULTIPLEXING AND ACCESS TECHNIQUES
Communications networks take great advantage of the concept of DAMA (Demand-Assigned Multiple Access). DAMA enables multiple devices to share access to the same network on a demand basis, that is, first come, first served. There exist a number of techniques for providing multiple access (i.e., access to multiple users) in a wireless network. Those techniques generally, but not always, are mutually exclusive.
11.5.1 Frequency Division Multiple Access
At the most basic level, frequency division is the starting point for all wireless communications because all communications within a given cell must be separated by frequency to avoid their mutual interference. Frequency Division Multiple Access (FDMA) divides the assigned frequency range into multiple frequency channels to support multiple conversations, as depicted in Figure 11.3. In other words, multiple narrowband frequency channels derive from a wider band of assigned radio spectrum, much as frequency division multiplexers (FDMs) operate in the wired world. Multiple incoming and outgoing calls contend for access to those channels. A given call takes place on one pair of frequencies, with one transmission in the forward direction and another for transmission in the reverse direction. At the same time, another call takes place on another pair of frequencies. The forward and reverse channels in each frequency pair are separated in frequency in order to avoid cochan-nel interference. The forward channels are separated from each other, as are the reverse channels, in order to minimize the potential for crosstalk. The station equipment must be frequency agile in order to search for and seize an available frequency channel, especially as the mobile transmitter/receiver moves from one cell to another in a cellular network.
Figure 11.3: Frequency division multiple access
Analog cellular systems employ FDMA. Advanced Mobile Phone System (AMPS), for example, provides for a total allocation of 40 MHz, which is divided into 666 (832 in some areas) frequency pairs. In each of the 734 serving areas defined by the FCC, the available 666 (or 832) channels initially were split equally between wireline (i.e., Incumbent Local Exchange Carrier, or ILEC) and nonwireline operators, with the carrier in each category initially having been determined on a lottery basis. AMPS separates the 30-kHz forward channel and the 30-kHz reverse channel in each frequency pair by approximately 55 MHz. On average, AMPS cell sites in the United States have a radius of approximately 1 mile. At maximum, based on a 7-cell reuse pattern, AMPS cells support about 56 frequency channels, considering that 21 of the available 416 channels per carrier are reserved for signaling and control purposes [5].
Frequency Division Duplex (FDD) is a means of providing duplex (bidirectional) communications. Forward and backward channels make use of separate frequencies. FDD is used with both analog and digital wireless technologies, including cordless telephony and cellular.
11.5.2 Time Division Multiple Access
Time Division Multiple Access (TDMA) is a digital technique that divides each frequency channel into multiple time slots, each of which supports an individual conversation (see Figure 11.4). This concept is exactly the same as in the wired world, where time division multiplexers perform the same function in a T/E-carrier environment. The total available bandwidth, the bandwidth of the individual channels, and the number of time slots per channel vary according to the particular standard in place, as well as the specific coding technique employed. GSM, for example, involves a carrier channel of 200 kHz, with a channel rate of approximately 200 kbps. The channel is divided into eight time slots of 25 kbps each, easily supporting low-bit-rate digitized voice of 9.6 kbps, plus overhead for framing and signaling. Each set of eight time slots is organized into a logical frame, and the frames are repeated frequently. Each conversation makes use of two time slots, one for the forward channel and one for the reverse channel. Assuming the same level of bandwidth, systems based on TDMA offer roughly three to four times the traffic capacity of those based on FDMA [1]. TDMA was specified first in EIA/TIA Interim Standard 54 (IS-54) and later was included in IS-136, which is an evolved version of IS-54 for use in cellular and PCS systems. TDMA is used in systems based on the D-AMPS and GSM cellular standards, and DECT cordless telephone standards.
Figure 11.4: Time division multiple access
A TDMA system, such as GSM, actually is both Frequency Division Multiplexing (FDM) and TDMA. FDM derives multiple carrier channels from a wider band of assigned spectrum. Within each frequency channel, TDMA derives multiple time slots (i.e., digital channels), for which incoming and outgoing calls contend.
E-TDMA (Enhanced TDMA), developed by Hughes Network Systems, is an improvement over TDMA, employing Digital Speech Interpolation (DSI) compression, also known as Voice Activity Detection (VAD), and half-rate vocoders (voice coders) operating at 4.8 kbps to enhance bandwidth utilization. E-TDMA provides as much as 16: 1 improvement over analog technology.
Time Division Duplex (TDD) is a digital means of providing bidirectional communications. TDD can be employed with both channels using the same frequency, but this ping-pong transmission approach can yield poor quality, as it actually is a half-duplex (HDX) transmission mode. (Note: In a properly engineered network, this HDX mode can be made transparent.) More commonly, TDD is used in conjunction with FDD, with the forward and backward TDM channels riding over separate frequency channels. Further, the time slots are staggered so the frequency-specific transceivers are not asked to transmit and receive at the same exact points in time. Time slot 1, for example, might be used on the forward channel, or uplink, from the user terminal to the centralized cell antenna, while time slot 3 is used on the backward channel, or downlink, from the cell site to the user terminal.
In addition to its enhanced traffic capacity when compared to FDMA, TDMA supports data traffic as well as voice. TDMA also offers considerable flexibility, as multiple time slots can be assigned to an application, depending on its bandwidth requirements.
11.5.3 Code Division Multiple Access
Code Division Multiple Access (CDMA) is a relatively new technology that has its roots in Spread-Spectrum (SS) radio. Hedy Lamarr, the famous actress and dancer of pre–World War II fame, created the concept of spread spectrum in 1940. As the story goes, Lamarr developed spread spectrum radio in order to remotely synchronize multiple player pianos. Droves of people supposedly paid good money to go to the resulting radio-controlled piano concerts in this much simpler time. The U.S. Patent and Trademark Office issued a patent to Ms. Lamarr and George Antheil, a film-score composer to whom she had turned for help in perfecting the idea, for a secret communication system that was, in effect, a spread spectrum radio. In the Pacific Theater during World War II, the Allies used that patented technology extensively to prevent the Japanese from jamming radio-controlled torpedoes. This primitive system used a mechanical switching system much like a piano roll to shift frequencies faster than the Nazis or the Japanese could follow them. Subsequently, spread spectrum has combined with digital technology for spy-proof and noise-resistant battlefield communications. During the 1962 Cuban nuclear missile crisis, for example, Sylvania installed it on U.S. warships sent to blockade Cuba, where the technology provided improved security as well as prevented signal jamming. Ms. Lamarr never asked for, and never received, any royalties from the use of her invention. Ms. Lamarr was quite an innovator, by the way, and across multiple disciplines. She delighted and shocked audiences in the 1930s by dancing in the nude in the movie Ecstacy [7].
As spread spectrum radio spreads the bandwidth of the transmitted signal over a spectrum of radio frequencies that is much wider than that required to support the native narrowband transmission, it commonly is known as a wideband radio technology. Multiple transmissions can occur at the same time in the same frequency domain. Thereby, multiple transmissions can each simultaneously use the entire system wideband, rather than just individual time slots or frequency channels. Spread spectrum uses two techniques: Direct Sequence (DS) and Frequency Hopping (FH).
· Direct-Sequence Spread Spectrum (DSSS) is a radio technique in which the narrowband signal is spread across a wider carrier frequency band. Each transmission is assigned a 10-bit pseudorandom binary code sequence, which comprises a series of 1s and 0s in a seemingly random pattern known to both the transmitter and receiver. The original code sequence is mathematically self-correlated to yield a code that stands out from all others, at least on average. The paired transmitters and receivers recognize their assigned and correlated code sequences, which look to all others as PseudoNoise (PN). DSSS phase modulates the carrier wave with a continuous string of PN code symbols, or chips, each of which has a much shorter duration than a data bit. So, the chip rate is much faster than the bit rate. Thereby, the noise signal occurs with a much higher frequency than the original data signal and spreads the signal energy over a much wider band (Figure 11.5). Rake receivers are used in CDMA systems to improve signal quality.
Figure 11.5: Code division multiple access
· Frequency-Hopping Spread Spectrum (FHSS), which generally is preferred over DSSS, more closely resembles Ms. Lamarr's original concept. FHSS involves the transmission of short bursts of data over a range of frequency channels within the wideband carrier, with the transmitter and receiver hopping from one frequency to another in a carefully choreographed hop sequence. This hop sequence generally is under the control of the centralized base station antenna. Each transmission dwells on a particular frequency for a very short period of time (e.g., no more than 400 ms for FCC-controlled applications), which may be less than the time interval required to transmit a single data packet, or symbol, or even a single bit. So, again, the chip rate can be faster than the bit rate. A large number of other transmissions also may share the same range of frequencies simultaneously, with each using a different hop sequence. The potential remains, however, for the overlapping of packets. The receiving device can distinguish each packet in a packet stream by reading the various codes prepended to the packet data transmissions and treating competing signals as noise.
CDMA improves bandwidth utilization because a great number of users can share the same wideband radio frequency channel (see Figure 11.5). While initial predictions were that CDMA would improve on AMPS by a factor of up to 20: 1, practical results have been more in the range of 15: 1 in cellular telephony applications. Particularly through the use of the FHSS approach, CDMA also provides excellent security, as it is virtually impossible to intercept more than a small portion of a transmission. Encryption, of course, can provide additional security [8]. FHSS also offers the advantage of improved overall transmission, since no individual transmission gets stuck with the assignment of a poor-quality channel.
Qualcomm perfected and commercialized CDMA and has gone on to develop, manufacture, market, and license CDMA products. The first commercial CDMA system was placed in service in Hong Kong, where cellular phones have long been considered a necessity and where the networks have suffered terrible congestion. Since then, a great number of manufacturers and providers of cellular, PCS, WLANs, and other systems and networks have licensed CDMA.
11.5.4 FDMA, TDMA, and CDMA Compared: It's Party Time!
The best commonsense means of comparing FDMA, TDMA, and CDMA is to use the cocktail party analogy, which has been used by acoustical engineers for many years. This is an international cocktail party with a great number of people all wanting to talk at the same time. The cocktail party is being held in a large ballroom (frequency band).
In an FDMA environment, each conversation takes place in its own space (channel). Therefore, the ballroom must be subdivided into smaller rooms (cells) of suitable size (bandwidth) and then subdivided into smaller cubicles (channels) so there is no interference between the pairs of people engaged in each conversation. At any given time during the conversation, each pair occupies a separate room. The pairs can move from room to room (cell to cell) as they make their way from the entrance to the buffet, although they must interrupt their conversations when they leave one room and must enter another empty room before resuming conversation (hard handoff, or break and make). The number of simultaneous conversations that can be supported depends on the number of rooms of suitable size that can be derived from the space available in the ballroom. The speakers also must control their volume and frequency levels in consideration of the thickness of the walls (guard bands, or frequency separations) so they do not interfere with others.
In a TDMA environment, multiple pairs of speakers may share an even smaller room, although they have to squeeze (compress) into the physical space. Six people, for example, can hold three separate conversations if they all take turns. The order of the conversations will be carefully controlled, as each pair will get a time slice (time slot) of 20s each minute (frame).
In a CDMA environment, the entire ballroom is open. A larger number of people can fit into the room because the walls have been removed. A larger number of pairs of people can engage in simultaneous conversations, with the conversations overlapping in frequency, volume, and time. Volume and frequency levels still must be controlled, of course. As highly intelligent receivers, the listening parties can distinguish the transmission of their paired speakers by locking on the language (PN code) spoken, even given the high level of background noise (interference) from other conversations. Since the walls have been removed, the pairs can move from one area to another without losing conversational connectivity (soft handoff, or make and break)
11.6 SPECIALIZED MOBILE RADIO
Two-way mobile radio dates to the very early days of radio. The U.S. Army Signal Corps mounted early spark transmitters in vehicles in 1904 and experimented with air-to-ground communications in 1908. The Detroit, Michigan, Police Department placed the first experimental one-way mobile radio dispatch system into service in 1921, operating in the 2-MHz band with the call letters KOP [9]. The first two-way mobile system was installed by the Bayonne, New Jersey, Police Department in the early 1930s. While this Amplitude Modulation (AM) radio application grew quickly, even as late as 1937 the FCC allocated only 40 channels to mobile radio. In the late 1930s, Frequency Modulation (FM) replaced AM as the method of choice because of its improved quality of reception and lower power requirements. (FM receivers tend to lock in on the stronger competing signal, whereas AM recognizes all competing signals.) In 1949, the FCC recognized two-way mobile radio as a new class of service and began to allocate more spectrum and to regulate its use [2, 10].
In 1946, AT&T was granted the first commercial license for two-way, mobile FM service. That first system in St. Louis, Missouri (United States), employed a centralized antenna with a range of 50 miles. The system not only served to interconnect mobile phones but also provided connection to the Public Switched Telephone Network (PSTN). As the service was fairly inexpensive and extremely convenient, it quickly grew in popularity and soon was oversubscribed. Similar systems were quickly deployed in other major U.S. cities, where they, too, were soon oversubscribed. In fact, it was not uncommon for a provider to load as many as 100 subscribers per channel, which resulted in horrible service quality. In 1976, for example, service in the New York metropolitan area consisted of 20 channels supporting 543 subscribers out of a total population of approximately 20 million. Not surprisingly, there was a waiting list of approximately 3700 [10].
Specialized Mobile Radio (SMR), also known as Trunk Mobile Radio (TMR), entered the scene in the 1960s, marketed as Improved Mobile Telephone Service (IMTS). This commercially available service made better use of FM bandwidth through narrowband communications involving smaller frequency channels. IMTS also enabled users to manually search multiple frequency channels. Shortly thereafter, intelligent mobile sets were developed that searched channels automatically. The concept of SMR/TMR remains much as it was originally. The provider places a radio tower and omnidirectional transmit/receive antennas on the highest possible point in the area and blasts the signal at the maximum allowable power level. As illustrated in Figure 11.6, this approach provides a coverage area of 50 miles or more, depending on topography. While some SMR systems support full-duplex (FDX) communications, many are only half-duplex (HDX). This HDX communications mode supports transmission in only one direction at a time and, therefore, requires that the parties take turns talking. The talker must depress a key or button on the microphone to talk and must release it to listen. This procedure is commonly known as the Push-To-Talk (PTT) protocol and sometimes is referred to as the press-to-talk or Citizens Band (CB) radio protocol. You copy, good buddy?
SMR/TMR largely has been supplanted by cellular service offerings, although it remains widely used in dispatch and fleet applications such as police, fire, and emergency vehicles as well as taxi fleets, utility fleets (e.g., telephone companies, gas and electric utilities, and CATV providers), and courier services. In the United States, 80 MHz has been allocated for SMR.
Enhanced Switched Mobile Radio (ESMR) is a technique developed by Nextel Communications and Geotek Communications for the development of a voice and data, cellular-like network using legacy SMR networks operating in the 800-and 900-MHz range. Nextel acquired and linked a large number of SMR networks throughout the United States, in February 1999 acquired the 191 900-MHz licenses of the bankrupt Geotek, and later added the 1.5-GHz band to the mix. Through the use of TDMA, each frequency channel is divided into multiple time slots to support multiple conversations. ESMR also supports call hand-off so mobile users can maintain connectivity as they travel from cell to cell. The Nextel network offers data throughput of 7.2 kbps, with coverage including most major metropolitan areas in the United States. Nextel terminal equipment supports integrated voice, data, paging, and Internet access. Perhaps the best known ESMR terminal is the Blackberry, a cellular telephone with features including Bluetooth technology, e-mail, PTT walkie talkie service, speakerphone, and Global Positioning System (GPS). The Blackberry is manufactured by Research in Motion (RIM) and the PTT service runs over the Nextel ESMR 800-MHz band. (Note: The Blackberry is by no means limited to the Nextel band.)
The walkie talkie was invented in 1938 by Al Gross, a high school student in Cleveland, Ohio (United States), at the time. The portable hand-held radio transmitter–receiver caught the attention of the Office of Strategic Services (OSS), predecessor to the Central Intelligence Agency (CIA). The OSS recruited Gross, who then led the effort to develop the walkie talkie for clandestine and military uses. Code named Joan/Eleanor, the first walkie talkie system comprised a ground unit, Joan, and an airborne unit, Eleanor. The system allowed OSS agents behind enemy lines to communicate with aircraft in a manner that virtually defied detection at the time. Gross also is credited with inventing the pager, the CB radio, and the cordless telephone. Gross also lobbied the FCC to create the Personal Radio license spectrum, which later became Citizens Radio Service Frequency Band, or CB radio. Gross formed the Citizens Radio Corporation, which manufactured and sold personal two-way radios, mostly to farmers and the U.S. Coast Guard [11].
11.7 PAGING
The paging system was invented by Al Gross as an adaptation of his two-way radio, the walkie talkie. After some early market resistance from doctors who were afraid that the system would upset their patients and disturb their golf games, Gross sold the first system in 1950 to New York's Jewish Hospital [11]. That first system provided a means by which a centralized antenna could broadcast alerts to small, inexpensive pagers, or beepers. A page simply transmitted an identification number, which was recognized only by the pager being addressed. If that pager were in range, it beeped, hence the term beeper. Response to the page was in the form of a telephone call to the paging company to retrieve a message. The FCC approved pagers for consumer use in 1958. The first consumer pager was the Motorola Pageboy I, which was based on the proprietary protocols including the GOLAY standard [10].
During the 1970s, an international team of radio engineers developed a standard set of code and signaling formats. That effort evolved into the POCSAG (Post Office Code Standardization Advisory Group) code, the name of which was derived from the fact that the British Post Office (BPO), which was the PTT for the United Kingdom at the time, chaired the effort. The POCSAG standard, which is in the public domain, provides for transmission speed of up to 2400 bps using channels of 25 kHz in the band 150–170 MHz. The CCIR (now ITU-R) standardized that code internationally in 1981, and most nations quickly adopted it. POCSAG can support as many as 2 million individual pager addresses. Tone-only, numeric, and alphanumeric pagers are supported on a one-way basis.
Paging in Europe has been constrained somewhat by the lack of agreement on common standards, although the POCSAG standard generally is recognized. A digital paging system known as ERMES (European Radio MEssage System) is supported by ETSI and the EU. In 1990, 26 system operators from 16 countries signed a Memorandum of Understanding (MoU) to create a pan-European system based on this standard. ERMES operates at 6250 bps in the band 169.4–169.8 MHz and uses Frequency Shift Keying (FSK) modulation.
Motorola more recently floated the FLEX set of proprietary solutions, which largely have replaced POCSAG in the United States and have become de facto standards throughout most of the world, excepting Western Europe. Those solutions provide two-way messaging, support data transmission, and provide greater bandwidth. FLEX also supports as many as 5 billion addresses, with up to 600,000 supported per channel. The FLEX family of protocols includes the following [12–15]:
· FLEX: 1600bps; 25-kHz channels; one-way
· ReFLEX: 1600, 3200, 6400, or 9600 bps; 25- or 50-kHz channels downstream and 12.5-kHz channel upstream; two-way
· InFLEXion: Up to 112 kbps; 50-kHz channels in the N-PCS (Narrowband PCS range); two-way; supports compressed voice downstream
Pagers generally operate over 25-kHz channels in the 900-MHz band. In 1984, the FCC (U.S.) dedicated 1 MHz of 40 channels in this band for nationwide paging purposes. RCCs (Radio Common Carriers) and PPOs (Private Paging Operators) provide paging services. Regulated by the FCC and the state Public Utility Commissions (PUCs), RCCs make use of FCC-designated frequencies. PPOs are unregulated but must share spectrum with other users in the VHF and UHF bands.
A typical page begins with a message transmitted to a centralized Network Operations Center (NOC). Using SkyTel, a Verizon company, as an example, the NOC forwards the page to a satellite with an appropriate footprint, or coverage area. As depicted in Figure 11.7, the satellite forwards the page to a terrestrial network of centralized antennas interconnected through various means, including private microwave, leased lines, and Frame Relay. The terrestrial antenna network forwards the page to the pager. In a two-way paging scenario, the response travels back to the terrestrial base station and then over a Frame Relay network to the NOC, where it is forwarded to the party who originated the page.
Figure 11.7: Terrestrial paging network with satellite interconnectivity
Generally speaking, these satellite links are highly beneficial and highly reliable. The May 1998 failure of the Galaxy IV satellite, however, underscored the vulnerability of even the most sophisticated networks. When the satellite lost its orientation, hundreds of terrestrial paging antennas had to be reprogrammed by hand in a process that took over a week. In addition to the 40 million or so pagers that lost service during this outage, 5400 of 7700 Chevron gas stations lost their pay-at-the-pump capabilities, and music-on-hold systems went down, causing callers to hang up when they thought they had been disconnected. Additionally, the results of the multistate Powerball lottery drawing could not be broadcast to the 88 TV stations that regularly carry that programming [16].
The downstream data to the pager originate in several ways. The most common approach involves the paging party dialing a telephone number which often is toll free. That number either may be dedicated to that one pager or may be one of many associated with the service provider. The telephone number terminates in a voice processor at the central location of the service provider. The voice processor prompts the paging party to use the touchtone keypad of the telephone set to enter a return number. If the telephone number is not dedicated to that one pager, the paging party first is prompted to enter a Pager Identification Number (PID), more generi-cally known as a Personal Identification Number (PIN), assigned to that one pager. If a textual message is to be sent, the paging service provider provides the option of accessing a human attendant who will answer the call and enter the message (generally limited to 500 characters) for transmission to the alphanumeric pager. Alternatively, many service providers now support Web-based messaging. This approach enables any paging party to access the service provider's website via the Internet and to enter a message of limited length without the intervention of an attendant. Additionally, large organizations with great numbers of employees equipped with pagers may have direct access to the paging system to send alphanumeric messages to their employees. Such access may be on either a dedicated or a dial-up basis and typically involves proprietary software which turns a PC into a paging dispatch terminal. These direct links from the user organization to the service provider also may permit the redirection of e-mail and facsimile transmission from corporate servers once those message formats are converted. As not to overwhelm both the paging network and the terminal equipment, filters must be used to limit the size of the transmitted file.
Pagers can be classified as tone only, numeric, alphanumeric, and voice enabled. Tone-only pagers cause the device to emit an audible tone and/or to vibrate or blink so it is not disruptive. Numeric pagers permit the receipt of numbers only on a display. The vast majority of contemporary pagers are alphanumeric, capable of receiving and displaying both alphabetic and numeric characters. Contemporary pagers contain enough memory to support as many as 30,000 characters. But relatively few pagers currently can support the storage of voice messages, which are extremely memory intensive. This voice capability is accomplished through the downloading of compressed voice mail from a centralized voice processor to the pager over a packet network.
Two-Way Paging (TWP) systems have been available abroad for years but were not introduced in the United States until roughly 1995. The simplest and most common version is known as 1.5-way paging. This approach supports guaranteed message delivery, as the network does not attempt to download messages until such time as the pager is within range, is turned on, and has enough memory to support the download. The general location of the pager is communicated upstream, so the messages can be downloaded to the antennas supporting that particular geographic area, rather than being broadcast across the entire paging network. Once downloaded successfully, the pager acknowledges to the network the receipt of the page. Such pagers are alphanumeric, supporting the display of a text message and return telephone number, typically for a telephone or pager. Full two-way paging enables the recipient of the page to select and transmit a return message, which most commonly is selected from a small set of predetermined messages, although some pagers support user-definable return messages.
Pagers also have been incorporated into other devices, such as cellular telephones, Personal Digital Assistants (PDAs), watches, and even key chains. Whether standalone or merged into another device, contemporary pager capabilities include some combination of alphanumeric display, two-way communications, message storage, audible and vibration alert, fax receipt (very unusual), and abbreviated e-mail forwarding (very unusual). Many paging service providers also offer information such as weather reports, sports scores, traffic reports, and stock quotes, all of which are provided through a feature known as Short Message Service (SMS), which currently is limited to 160 characters [17]. Limited Web content also may be available. Incoming messages generally may be reviewed, erased, and archived. Both incoming and outgoing messages may be date and timestamped.
11.7.3 Paging Applications: Contemporary and Developing
Pagers reached peak popularity in the 1980s and 1990s. In 1994, there were estimated to be well over 61 million pager subscribers worldwide, according to some estimates [18]. In the United States alone, there were over 27 million subscribers to over 2000 paging services [19]. At that point, pagers were overwhelmed by cellular telephones, which offer much more functionality, although at considerably higher cost. As a result, millions of pagers have been disconnected over the past decade or so. Pagers remain used in areas where cellular service is not available and in applications where cellular functionality is undesirable from a cost standpoint.
Paging networks have found new life in a host of applications that were unimag-ined only a few years ago. CreataLink paging service from Motorola takes paging to the contemporary extreme, in the Americas at least. Through dialing a toll-free number, you can access a menu of eight vehicle-related functions that include turning the lights off and on, vehicle tracking, starting or disabling the engine, locking or unlocking the doors, and opening or closing the trunk. Similarly, into the future, you can unlock the house so that your children have access to the home after school when you are at work. Two-way pagers send information from patients with medical emergencies and receive authorization to automatically increase the dosage of medication from an implanted drug-release system. Two-way pagers also are used in telemetry applications for network-based remote utility meter reading. Tank monitoring supports the remote monitoring of content levels and material corrosion. Lighting control, irrigation control, climate control, and remote monitoring of vending machine inventories are other applications [20]. The Motorola ReFLEX CreataLink transceivers transmit in the range 896–902 MHz at speeds up to 9600 bps and receive in the range 929–942 MHz at speeds up to 6400 bps [21, 22].
Paging costs generally involve a flat monthly fee which includes some number of pages. The cost of pages above the threshold generally is sensitive to their nature (e.g., numeric, alphanumeric, or voice). A few paging service providers offer the option of paging party pays, which charges the paging party, rather than the paged party, for the page.
11.8 CORDLESS TELEPHONY AND WIRELESS OFFICE TELECOMMUNICATIONS SYSTEMS
Cordless telephones are quite simply telephone handsets that connect on a wireless RF basis to a base station that connects to a wall jack via an RJ11 or other standard jack. Wireless Office Telecommunications Systems generally are in the form of adjuncts that provide cordless telephony communications capabilities behind PBXs, Electronic Key Telephone Systems (EKTS), hybrid's KTS, or Centrex systems. They generally are limited to voice applications, although some also support low-speed data. WOTS involve a wireless master controller, which is hardwired to special ports on the PBX, KTS, Hybrid, or Centrex. As illustrated in Figure 11.8, the master controller is hardwired to subcontrollers and antennas, which are distributed throughout the office complex or campus in a picocell configuration. The terminal equipment is in the form of wireless handsets, which are generally low in cost and limited in range. Some dual-mode handsets exist, which can function as traditional cellular phones when the user is out of range of the WOTS or when otherwise desired.
Figure 11.8: WOTS configuration
Generally an extension and application of common cordless telephony, WOTS provides the advantage of mobility for a small group of select employees who must have the freedom to wander around the complex but must have communications capability at all times. However, WOTS PBX adjuncts tend to be highly capacity limited and costs can be easily double those of traditional phones when the additional infrastructure costs are taken into consideration. WOTS falls into the general category of low-tier systems, which are intended for pedestrian, in-building, and Wireless Local Loop (WLL) application. There exist multiple cordless telephony standards used to provide WOTS service. The original cordless telephones (c. 1980) in the United States were assigned 1 of 10 channels in the 27-MHz range. As these early analog phones transmitted in the clear, that is, with no encryption or signal-scrambling mechanism, and as there were so few channels, it was not unusual to accidentally engage in a three-way conversation with a neighbor who happened to have a phone operating on the same channel. In 1986, the FCC changed the cordless frequency range to the 46-and 49-MHz bands and reduced the allowable power levels, which had the effect of reducing the range slightly but improving signal quality. Contemporary digital versions operate in the 900-MHz and 2.4-GHz bands, which are in the unlicensed Industrial/Scientific/Medical (ISM) band. Contemporary standards include the following, the most significant of which are compared in Table 11.1:
· CT1 (Cordless Telephony generation 1) was developed in Europe, where it was known as CEPT-1. CT1 operates in the 915-and 960-MHz bands over 40 paired channels 25 kHz wide. FDMA and FDD derive two separate channels, one for transmission and one for reception, each of which is 12.5 kHz wide. CT1 is analog and low cost but limited in range to 150m or so. FM was employed in this voice-only technology. A variation on this standard is CT0, which was primarily used in the United Kingdom. CT0 specified eight paired channels, with the base station transmission in the 1.642–1.782-GHz range and portable station transmission in the 47-MHz range. A number of parochial CT0 versions were developed in other countries.
· CT1 + is a variation on CT1 developed in concert by Belgium, Germany, and Switzerland. CT1+ was intended as the basis for a public wireless service, along the lines of Telepoint. CT1+ operated in the 887-and 932-MHz bands over 80 channels 25 kHz wide. FDMA and FDD derive two separate channels, one for transmission and one for reception, each of which is 12.5 kHz wide. Although CT1+ was not successful, it did originate the concept of a Common Air Interface (CAI), which enables multiple manufacturers to develop products in support of a public cordless telephony service offering.
· CT2 was developed in the United Kingdom, where it formed the technology basis for the ill-fated Telepoint public cordless service. CT2 is a digital technology using TDMA and TDD and is deployed on a limited basis in Europe, Canada, and the Asia-Pacific. While it originally supported only outgoing calling, contemporary CT2 implementations support two-way calling. As CT2 does not support hand-off, the user must remain within range of the antenna used to set up the call. CT2 operates in the range 864–868 MHz, supports 40 channels spaced at 100 kHz, and uses Gaussian Frequency Shift Keying (GFSK) as the modulation technique. Dynamic channel allocation requires a frequency-agile handset. CT2 was the first international standard providing a Common Air Interface (CAI) for systems operating in the 800-and 900-MHz bands. CT2 supports data communications at rates up to 72 kbps.
· CT2 + is an improvement on CT2, supporting two-way calling and call hand-off. CT2+ uses 8 MHz of bandwidth in the 900-MHz range. CT2+ is based on dynamic channel allocation, requiring frequency-agile handsets. Encryption is supported for improved security. A common signaling and control channel offers improved call setup times, increased traffic capacity, and longer battery life because the handset must monitor only the signaling channel. CT2+ has been used in applications such as the Walkabout public cordless telephony trial in Canberra, Australia.
· CT3 was developed by Ericsson in 1990 as a proprietary solution designed for high-density office environments. CT3 is based on TDMA and TDD and uses the same frequency bands as CT2+, supporting roaming and seamless hand-off. CT3 supports Telepoint (PCS) and in-building office applications (WOTS).
· Digital Enhanced (nee European) Cordless Telecommunications (DECT) is the pan-European standard for digital cordless telephony using TDMA and TDD. Ratified by ETSI in 1992, DECT provides 10 FDM channels in the band 1880–1990 MHz. Channel spacing is at 1.728 MHz, and each channel will support 1.152Mbps with Gaussian Frequency Shift Keying (GFSK) as the modulation technique. Each channel supports 12 users through TDMA, for a total system load of 120 users. Voice encoding is Adaptive Differential Pulse Code Modulation (ADPCM) at 32 kbps. As DECT supports handover, users can roam from cell to cell as long as they remain within range of the system. DECT antennas can be equipped with optional spatial diversity to deal with multipath fading. Security is provided through authentication and encryption mechanisms. In North America, DECT is the basis for the Personal Wireless Telecommunications (PWT) standard, which operates in the unlicensed band 1910–1920 MHz. PWT/E is an extension into the licensed bands 1850–1910 and 1930–1990 MHz [25].
· Personal Handyphone System (PHS) is a digital cordless telephony system developed in Japan and used in China, Taiwan, and throughout Asia. PHS carves out 77 carriers with spacing of 300 kHz in the band 1895–1918.1 MHz. The upper half of the band is used for public systems and the lower half for home office applications. Each carrier supports four TDMA channels employing π/4 Differential Quaternary Phase Shift Keying (π /4 DQPSK) to yield a theoretical data rate of 384 kbps. Voice is encoded at 32 kbps. PHS enjoyed considerable market success but lately has declined in the face of competition from low-cost cellular offerings. (Note: π is the Greek letter pi, which is the symbol for the ratio of the circumference of a circle divided by its diameter. Pi is a mathematical constant with an approximate value of 3.14159. Pi is from the Greek periphereia, which translates into periphery, i.e., circumference.)
· Personal Communications Services (PCS) in the United States is based on the Wireless Access Communication System (WACS), which subsequently was modified to an industry standard known as Personal Access Communications Services (PACS). PCS employs FDD to carve out 16 paired downstream and upstream carriers with spacing of 300 kHz in the bands 1850–1910 and 1930–1990 MHz. Within each frequency channel, PCS derives eight TDMA channels employing π /4 Quadrature Phase Shift Keying (π /4 QPSK). Voice encoding is ADPCM at 32 kbps and the theoretical data rate is 384 kbps. PCS has never gained any traction in the United States due largely to the prevalence and low cost of cellular telephony [23, 24, 26, 27].
|
Table 11.1: Cordless Telephony Standards Open table as spreadsheet |
|||||||
|
Standard |
CT1 |
CT1+ |
CT2 |
CT2+ |
DECT |
PHS |
PCS |
|
Region |
Europe |
Europe |
Europe |
Europe |
Europe |
Japan |
U.S. |
|
Frequency band, MHz |
915/960 |
887/932 |
864/868 |
944/948 |
1880-1990 |
1895-1907 |
1850-1910,1930-1990 |
|
Carrier spacing |
25 kHz |
25 kHz |
100 kHz |
100 kHz |
1.728 MHz |
300 kHz |
300 kHz |
|
Number of FDM carriers |
40 |
80 |
40 |
40 |
10 |
75 |
32 |
|
Channels/carrier |
1 |
1 |
1 |
1 |
12 |
4 |
8 |
|
Access method |
FDMA |
FDMA |
TDMA |
TDMA |
TDMA |
TDMA |
TDMA |
|
Duplex method |
FDD |
FDD |
TDD |
TDD |
TDD |
TDD |
FDD |
|
N/A |
N/A |
72kbps |
72 kbps |
1.152 Mbps |
384 kbps |
384 kbps |
|
|
Modulation method |
FM |
FM |
GFSK |
GFSK |
GFSK |
π/4 DQPSK |
π/4 DQPSK |
|
Speech coding, kbps |
N/A |
N/A |
32 |
32 |
32 |
32 |
32 |
|
|||||||
|
Source: [23, 24]. |
|||||||
As I discuss in Chapter 3, contemporary PBXs increasingly are Internet Protocol (IP) based, running on a switched Ethernet LAN platform. Mirroring the trend toward WLANs, there currently is a great deal of interest in Voice over Wi-Fi (VoWiFi), which I discuss in Chapter 8. This interest level is heightened in the context of the developing 802.11n recommendation, which includes MIMO antenna technology. Once that standard matures and multimode VoWiFi/cellular handsets become available at reasonable cost, VoWiFi may very well gain the market acceptance that seems to have eluded WOTS cordless telephony.
11.9 CELLULAR RADIO
The basic concept of cellular radio dates back to 1947, when numerous, low-power transmit/receive antennas were scattered throughout a metropolitan area to increase the effective subscriber capacity of SMR/TMR radio systems. This architecture broke the macrocell area of coverage into smaller cells. Thereby, nonadjacent cells could reuse each frequency. Additionally, the carrier could split, or subdivide, the resulting cells further as the traffic demands of the system increased. This cellular concept is highly scaleable. Traditional cellular radio, I should note, involves a circuit-switching mode.
This original concept is the basis of cellular radio. The first prototype system was developed by AT&T and Bell Labs in 1977. In 1979 in Tokyo, Nippon Telephone and Telegraph (NTT) activated the first commercial cellular system, utilizing 600 duplex analog radio channels in the band 925–940 MHz for the uplink channels from the Mobile Station (MS) to the Base Station (BS) and the band 870–885 MHz for the downlink, or reverse, channels [26]. The first system in the United States was activated in Chicago on October 13, 1983. AT&T operated that AMPS network for exactly 79 days, at which point the Modified Final Judgment (MFJ) took effect and a subsidiary of Ameritech (now part of SBC) assumed ownership [2]. By 1984, the Chicago network already was saturated in some cells and cellular telephones were in the hands of 91,600 people, growing to 19 million by 1994 [2, 3]. A cellular telephony network comprises multiple low-power transmit/receive antennas distributed throughout a geographic area, with each cell site having a relatively small, more or less circular area of coverage (see Figure 11.9). The coverage area of each individual cell overlaps those of neighboring cells, with the cell diameter generally a minimum of about 1 mile and a maximum of about 5 miles, sensitive to factors such as topography and traffic density [8]. As the terminal device moves out of the effective range of one cell, the call switches from one cell antenna to another through a process known as hand-off in order to maintain connectivity at acceptable signal strength. The hand-off is controlled through a Mobile Telephone Switching Office (MTSO), which is the functional equivalent of the PSTN Central Office (CO). The MTSOs generally are interconnected and connected to the PSTN through either private microwave or leased-line facilities.
Figure 11.9: Cellular network with MTSO connected to PSTN
The process of hand-off can be accomplished in several ways. The break-and-make approach, also known as a hard hand-off, breaks the connection with one cell site antenna before connection is reestablished with another preselected cell site. As the duration of the break is very short, it is not noticeable in voice communication. However, it renders data communications difficult at best. Make-and-break, also called soft hand-off, makes the new connection before breaking the old. This gentler approach offers considerable advantages in the transmission of data, par-ticularly when supporting high-speed vehicular traffic, which may move between many cells in a relatively short period of time. Cellular systems, which support high-speed traffic, generally are known as high-tier systems. Low-tier systems include those intended for pedestrian traffic or for in-building or WLL applications.
Each cell site supports a limited number of frequency channels in order to take advantage of frequency reuse. The U.S. analog AMPS networks, for example, divide 333 frequencies (416 in certain areas) per carrier among cell sites; the average cell site supports 56 channels. To improve the performance of the network by reducing crosstalk, the original omnidirectional antennas were replaced with vectored versions. Vectoring generally involves three vectors, each with coverage of 120 °. Splitting the frequencies into three vectors had the unfortunate effect of reducing by two-thirds the number of channels available to any given user in the cell site. As a result, more cell sites of smaller size were required, which entailed additional cost. For aesthetic reasons, however, many cities and towns placed moratoria on the construction of new antenna sites. In response, antennas were developed that can support the full 333/416 channels, subdividing the area of coverage through the use of 12 beams, each with a 360 ° sweep. These smart antennas communicate continuously with the MTSO so that channel allocation is managed cell by cell and on the basis of the network as a whole and in consideration of shifts in traffic patterns [16].
11.9.1 Cellular Standards
Cellular standards are numerous and largely incompatible. Standards include both early analog and more recent digital solutions. Digital systems offer the advantages of improved error performance, improved bandwidth utilization through compression, and enhanced security through encryption and other mechanisms. Digital systems also support data communications much more effectively. In the United States, carriers are shifting subscribers to digital systems through various marketing enticements. The FCC has authorized U.S. carriers to cease support for analog cellular systems as of March 1, 2008.
11.9.1.1 Generation 1 (1G): Analog Cellular
The first cellular systems were analog in nature. Collectively, these analog solutions are categorized as 1G (1st Generation) systems. They include the following and are compared side by side in Table 11.2.
|
Table 11.2: Analog Cellular Standards Open table as spreadsheet |
|||||
|
Standard |
AMPS |
ETACS |
NTACs |
NMT 450 |
NMT 900 |
|
Region |
U.S. |
UK |
UK |
Nordic |
Nordic |
|
Frequency band, MHz |
Tx: 824-849 |
Tx: 871-904 |
Tx: 915-925 |
Tx: 453–458 |
Tx: 890-915 |
|
|
Rx: 869-894 |
Rx: 916-949 |
Rx: 860-870 |
Rx: 463–468 |
Rx: 935-960 |
|
Carrier spacing, kHz |
30 |
25 |
12.5 |
25 |
12.5 |
|
Number of FDM carriers |
666/832 |
1000 |
400 |
200 |
1999 |
|
Channels/carrier |
1 |
1 |
1 |
1 |
1 |
|
Access method |
FDMA |
FDMA |
FDMA |
FDMA |
FDMA |
|
Duplex method |
FDD |
FDD |
FDD |
FDD |
FDD |
|
N/A |
N/A |
N/A |
N/A |
N/A |
|
|
Modulation method |
FM |
FM |
FM |
FM |
FM |
|
|||||
11.9.1.1.1 Advanced Mobile Phone System (AMPS)
AMPS was the first cellular technology deployed in the United States. Developed by Motorola and AT&T, AMPS is an analog technology operating on 50 MHz in the 800-MHz band and supporting 666 (in some areas 832) channels. In the United States, 25 MHz and 333 (in some areas 416) channels each are provided to the A-carrier, or nonwireline carrier, and the B-carrier, or wireline carrier (incumbent telco or telco consortium). Of the total number of channels awarded to each carrier, 21 channels are non-conversational channels dedicated to call setup, call hand-off, and call teardown. The remaining communications channels are split into 30-kHz voice channels, with separation of 45 MHz between the forward and reverse channels. Based on FDMA and FDD transmission, AMPS does not handle data well, with modem transmission generally limited to 6.8 kbps. Although once widely deployed in the United States, Australia, the Philippines, and other countries, AMPS has almost entirely been replaced by digital technology. Australian regulators mandated a cutover from analogue (That's Aussie for analog, mate.) AMPS to digital GSM and CDMA beginning December 31, 1999, in Melbourne, gradually extending throughout the country during 2000. As noted above, the FCC in the United States has authorized carriers to cease support for analog systems as of March 1, 2008.
11.9.1.1.2 Narrowband AMPS (N-AMPS)
Also developed by Motorola, N-AMPS enhances the performance of an analog AMPS system. System capacity is improved by splitting each 30-kHz channel into three 10-kHz channels, thereby tripling AMPS capacity. Very few U.S. carriers deployed N-AMPS.
11.9.1.1.3 Total Access Communications System (TACS)
TACS is a derivative of AMPS developed for use in the United Kingdom in the 900-MHz band. TACS supports either 600 or 1000 channels, each of 25 kHz, compared with the 666/832 channels supported by AMPS. A number of variations were developed, including Narrowband TACS (NTACS), Extended TACS (ETACS), and Japanese Total Access Communications System (JTACS). TACS found acceptance in very few nations, largely has been replaced by GSM, and is considered obsolete in the United Kingdom.
11.9.1.1.4 Nordic Mobile Telephone (NMT)
NMT was developed and placed into service in the early 1980s in Scandinavian countries, including Denmark, Finland, Norway, and Sweden. NMT 450 operates in the 450-MHz range, which yields excellent signal propagation. Therefore, it is especially appropriate for sparsely populated areas supported by few cell sites. NMT 450 found little acceptance outside of the Scandinavian countries. NMT 900 operates in the 900-MHz range and is appropriate for more densely populated areas. NMT 900 found acceptance in certain countries in Asia as well as the Nordic countries, although it is not considered a long-term technology. NMT largely has been replaced by GSM [23, 26, 27].
11.9.1.2 Generation 2 (2G): Digital Cellular
Digital cellular clearly dominates the cellular radio world, having almost completely replaced the analog systems. There are a great number of standards, and a number of standards-based solutions, none of which are compatible. Collectively, these systems are grouped into the 2G (2nd Generation) category. They include the following, the most important of which are compared side by side in Table 11.3.
|
Table 11.3: Digital Cellular Standards Open table as spreadsheet |
||||
|
Standard |
IS-54/136 (D-AMPS) |
IS-95 (cdmaOne) |
GSM |
PDC |
|
Region |
North America |
North America |
International |
Japan |
|
Frequency band, MHz |
Tx: 824-849 |
Tx: 824-849 |
Tx: 890-915 |
Tx: 940-956,1477-1501 |
|
|
Rx: 869-894 |
Rx: 869-894 |
Rx: 935-960 |
Rx: 810-826,1429-1453 |
|
Carrier spacing, kHz |
30 |
1250 |
200 |
25 |
|
Number of FDM carriers |
832 |
20 |
124 |
1600 |
|
Channels/carrier |
3 |
798 |
8 |
3 |
|
Access method |
TDMA |
CDMA |
TDMA |
TDMA |
|
Duplex method |
FDD |
FDD |
FDD |
FDD |
|
48.6 kbps |
1.288 Mbps |
270.833 kbps |
42 kbps |
|
|
Modulation method |
π /4 DQPSK |
BPSK/OQPSK |
GMSK |
π /4 DQPSK |
|
Speech coding |
VSELP: 8 kbps |
EVRC: 13 kbps |
RPELPC: 13 kbps |
VSELP: 9.6 kbps |
|
|
|
CELP: 8 kbps |
VSELP: 8 kbps |
CELP: 5.6 kbps |
|
||||
11.9.1.2.1 Digital-AMPS (D-AMPS)
Also known as US TDMA and NA-TDMA (North American TDMA), D-AMPS was specified in IS-54 and later evolved into IS-136. D-AMPS is a North American digital cellular standard that operates in the same 800-MHz band as the earlier analog AMPS. In fact, the two can coexist in the same network. D-AMPS uses the same 30-kHz bands as AMPS and supports up to 416 frequency channels per carrier. Through Time Division Multiplexing (TDM), each frequency channel is subdivided into six time slots, each of which operates at 8 kbps. Each call initially uses two time slots (e.g., 1 and 4, 2 and 5, and 3 and 6) in each direction, for a total of 16 kbps, which supports the data transfer plus overhead for call processing. While the standard recommends speech compression at 8 kbps (actually 7.95 kbps) using Vector-Sum Excited Linear Predictive Coding (VSELP), that is an average rate because each call can burst up to 48 kbps. D-AMPS yields a 3: 1 advantage over AMPS in terms of bandwidth utilization. IS-136 is known as a dual-mode standard because both D-AMPS and AMPS can coexist on the same network, with both using the same 21 control channels for call setup, call hand-off, and call teardown. Thereby, IS-136 offers carriers the advantage of a graceful transition from analog to digital. IS-136 also includes a nonintrusive Digital Control Channel (DCCH), which is used for Short Message Service (SMS) and caller ID. SMS supports information transfer for applications such as weather reports, sports scores, traffic reports, and stock quotes, as well as short e-mail-like messages, which may be entered through the service provider's website. Some service providers also allow the cellular user to respond to e-mails via two-way SMS. Data communications is supported at up to 9.6 kbps per channel (paired time slots), and as many as three channels can be aggregated for speeds up to 28.8 kbps. Group 3 facsimile also can be supported. The RF modulation technique is DQPSK (Differential Quaternary Phase Shift Keying). Cingular Wireless uses D-AMPS.
11.9.1.2.2 Global System for Mobile Communications (GSM)
Originally known as Groupe Spéciale Mobile (French), GSM was adopted by the CEPT in 1987 as the standard for pan-European cellular systems and was first introduced in 1991. GSM operates in the 800-MHz and 900-MHz frequency bands and is ISDN compatible. GSM carves each 200-kHz band into eight TDMA channels of 33.8 kbps, each of which supports a voice call at 13 kbps using Linear Predictive Coding (LPC). Data throughput generally is limited to 9.6 kbps, due to FEC and encryption overhead. GSM commonly employs a four-cell reuse plan, rather than the seven-cell plan used in AMPS, and divides each cell into 12 sectors. GSM commonly uses frequency hopping and time-slot hopping, which also is used in CDMA systems. GSM offers additional security in the form of a Subscriber Identification Module (SIM), which plugs into a card slot in the handset, much as a PCMCIA card fits into a laptop computer. The SIM contains user profile data, a description of access privileges and features, and identification of the cellular carrier that hosts the home registry. The SIM can be used with any GSM set, thereby providing complete mobility across nations and carriers supporting GSM, assuming that cross-billing relationships are in place. GSM clearly developed to be the international standard of choice. Like D-AMPS, GSM supports SMS text messaging, which generally is two-way. GSM is in place in over 475 networks in more than 190 countries and predominates throughout Europe and much of Asia, supporting full roaming privileges from country to country. With minor modifications, GSM is the basis for DCS 1800, also known as PCN (Personal Communications Network), in Europe. DCS 1800, in large part, is an upbanded version of GSM, operating in the 1800-MHz (1.8-GHz) range. Also with minor modifications, it is the basis for PCS 1900 in the United States, where it also is known as GSM. PCS 1900 is the ANSI standard (J-STD-007, 1995) for PCS at 1900 MHz (1.9 GHz). Unfortunately, PCS 1900 is not compatible with the original European GSM, due to the difference in frequency bands. T-Mobile (owned by Deutsche Telekom, which explains a lot) has deployed PCS 1900, and Cingular built a GSM network as an overlay to its D-AMPS network.
11.9.1.2.3 Personal Communications System (PCS)
PCS is a U.S. term for cellular systems based on EIA/TIA IS-95a, also known as cdmaOne and CDMA Digital Cellular. The first commercial systems were installed in South Korea and Hong Kong in 1995 and in the United States in 1996. PCS uses one or more frequency bands of 1.25-MHz converted from the existing AMPS spectrum of each carrier deploying the service. Each 1.25-MHz band is subdivided into 20 carriers, each of which can support as many as 798 simultaneous calls and aggregate bandwidth of up to 1.288 Mbps. Because CDMA is employed, no guard bands are required. IS-95 can support dual-mode communications, operating in the same network as AMPS. As previously noted, CDMA offers the advantages of improved bandwidth utilization as compared to AMPS (as much as 10: 1 or even 20: 1) and TDMA (as much as 6: 1), soft hand-off, variable-rate speech-encoding, and support for both voice and data. The basic user channel rate is 9.6 kbps, although various channel rates can be achieved depending on the carrier implementation. The variable-rate speech-encoding algorithm runs at maximum rates of 8 kbps using Code-Excited Linear Prediction (CELP) or 13 kbps using Enhanced Variable-Rate voCoder (EVRC) and varies the rate downward to as low as one-eighth rate if the level of speech activity permits. The IS-95-B specification supports symmetric data rates of 4.8 and 14.4 kbps per channel; as many as eight channels can be aggregated to support a data rate up to 115.2 kbps. CDMA offers additional advantages in terms of maximum cell size due to improved antennas sensitivity and battery time due to precise power control mechanisms. IS-95 has been deployed by Airtouch (now part of Verizon), Ameritech Cellular (now part of Cingular), AT&T Wireless (now part of Cingular), Bell Atlantic Mobile (now part of Verizon), GTE MobilNet (now part of Verizon), 360 ° Communications (now part of ALLTEL), and Sprint Nextel [28–31].
11.9.1.2.4 Personal Digital Cellular (PDC)
Previously known as Japanese Digital Cellular (JDC), PDC is a Japanese standard for a digital system operating in the 800-, 900-MHz, and 1400-MHz frequency ranges. PDC derives 1600 RF carriers, each of which supports three TDMA channels at 42 kbps. Full-rate voice encoding is at 9.6 kbps and half-rate at 5.6 kbps. PDC has not found acceptance outside Japan, where it was extremely popular, with the exception of a few Asian countries under Japanese economic influence. PDC is being phased out in Japan, in favor of cdma2000.
11.9.1.3 Cellular Data Communications: The Early Years
Mobile data communications has always been somewhat difficult at best. The first attempt at mobile datacomm, at least as best I can determine, dates to 1907, when the U.S. Cavalry experimented with various solutions to a particular tactical field communications problem. When a cavalry troop was on patrol, it required some means of communicating back to a command-and-control center. At the time, therefore, a single mounted soldier trailed behind the troop, unreeling copper wire from a small wagon as he rode along. When it was necessary to send a message, the troop either stopped until the communications specialist caught up or sent a rider back to meet him. Then the communications specialist reined in his mount, hopped off, planted a metal stake in the ground to complete the circuit, and sent a telegraph message via Morse code to the company command center. This approach clearly slowed the movement of the troop, placing it at a tactical disadvantage. Some brilliant engineers devised a solution which involved shaving a spot on the horse's rump and gluing a copper patch to the horse's bare skin. A telegraph key was mounted on the copper patch, which was connected to the copper wire that connected back to the command center. Because a horse almost always has at least one foot on the ground, the circuit remained complete and the communications specialist could send a message while riding [32]. No mention is made of the attitude of the horse toward this technological breakthrough. Neither is any mention made of the bit error rate (pun intended). In any case, the technique apparently proved less than satisfactory, as it merited only a few references in obscure military documents. Wireless technology certainly offered greater promise.
The 1G analog networks really were not designed with data communications in mind. Over time, a number of manufacturers developed modems that could achieve theoretical data rates of as much as 28.8 kbps under optimum conditions, but conditions were rarely optimum. Factors affecting signal quality include the distance between the mobile station and the base station and Multi-Path Interference (MPI). Given the high error rates typical in cellular networks, the modem protocol either must include Forward Error Correction (FEC), which is overhead intensive, or must provide for repeated transmissions of errored and lost data packets or both. All things considered, throughput generally was more in the range of half the advertised rate at best. As analog systems take a break-and-make approach to call hand-off between cells, dropped connections were routine for the mobile cellular data user.
The 2G digital cellular systems improved on the analog approach, offering all digital data communications advantages enjoyed in the wired world. Those advantages include more usable bandwidth, improved error performance, and therefore enhanced throughput. Issues of inherent error performance and MPI remain, however. The hard hand-offs associated with the break-and-make connection technique persisted in 2G networks, with the exception of GSM, which used the softer approach of make-and-break to maintain connectivity.
In support of data communications over cellular networks, several manufacturers unveiled data-ready cellular phones as early as January 1995. Display screens varied in size from 16 to 160 characters. In addition to voice and SMS, those early data-ready phones supported Internet access, facsimile, and e-mail [33]. Nonetheless, the early cellular networks were not particularly friendly toward data communications.
Cellular Digital Packet Data (CDPD) was formalized in 1993 by a consortium of carriers in the United States to resolve this issue. CDPD operated over existing AMPS networks in the 800-MHz band, taking advantage of the natural idleness in cellular networks in between disconnections and connections and during the break-and-make process to transmit packetized data at rates up to 19.2 kbps using either the Internet Protocol (IP) or the ISO Connectionless Network Protocol (CLNP). The frequency-agile CDPD modems searched for available channels over which to send encrypted packets during the periods of channel idleness. As this approach proved expensive, however, the carriers ultimately deployed CDPD over AMPS channels removed from voice service as analog cellular voice subscribers gradually transitioned to digital service. Ultimately, CDPD proved too difficult, too bandwidth limited, and too expensive, especially as data-ready 2.5G and 3G networks made their appearances. The last of the CDPD networks were finally decommissioned around the end of 2005.
11.9.1.4 Generation 2.5 (2.5G) and 3 (3G) and beyond
Beginning in 1992, the ITU-R began work on International Mobile Telecommunications-2000 (IMT-2000). Variously dubbed 2.5G (i.e., the generation midway between 2G and 3G) and 3G, some of the many standards are well developed and widely available while others require a bit more work. They all grew out of an attempt to define a single global standard, which effort collapsed in October 1999, when representatives from the various countries agreed to adopt federal standards under the IMT-2000 umbrella. Two of the three modes are based on Code Division Multiple Access (CDMA) and one on Time Division Multiple Access (TDMA). cdmaOne, also known as Telecommunications Industry Association Interim Standard-95a (TIA IS-95a), was the first CDMA-based 2G approach to be introduced and is popular with CDMA-based cellular operators in North America and Asia. CDMA2000 3XMC, the high-speed version operating at 2 Mbps, was developed by Qualcomm and has been approved by the ITU-R. Enhanced Data Service for GSM Evolution (EDGE) is the TDMA variant of IMT-2000. General Packet Radio Service (GPRS), an interim step toward EDGE, is the choice of cellular operators with networks based on Global System for Mobile Communications (GSM) and Digital Advanced Mobile Phone Service (D-AMPS). Regardless of the transition approach and the specifics of the technology, all 3G systems will be based on CDMA.
In general, these standards support much increased data rates for both mobile and fixed wireless versions. In general, each has its roots in and is an extension of a 2G technology. So, the incompatibilities roll on, although at faster speeds, in support of voice, Short Message Service (SMS) and Multimedia Messaging Service (MMS), and Internet access. The 2.5G networks run over the same spectrum as the 2G networks and are backward compatible with them. Hence the transmission rates are lower than those of 3G networks, which run the over the spectrum set aside for their specific purposes. 2.5G and 3G technologies and standards include EDGE, GPRS, IMT-2000, UMTS, and W-CDMA. All of these approaches fit under the umbrella of International Mobile Telecommunications-2000 (IMT-2000), which replaced Future Public Land Mobile Telecommunications System (FPLMTS) as the vision for a single global standard for wireless networks. IMT-2000 is an ITU initiative for a twenty-first century wireless network architecture. As illustrated in Figure 11.10 and compared in Table 11.4, specifications include 128/144 kbps for high-mobility applications, 384 kbps for pedestrian speed (i.e., walking speed) applications, and 2.048 Mbps for both fixed WLL (Wireless Local Loop) and in-building applications such as WLANs (Wireless LANs). IMT-2000 also was intended to operate in the 2-GHz band. Note that 2000 has several meanings: the year 2000, bandwidth of up to 2000 kbps, and frequency range 2000 MHz. As it turned out, the year 2000 was pushed back to at least 2002, bandwidth of 2 Mbps is only for limited in-building applications, and the 2-GHz spectrum has been allocated for other pur-poses in North America as part of the unlicensed ISM band.
|
Application |
Maximum Bit Rate |
Cell Size |
|
Mobile |
Macro/micro/pico |
|
|
Pedestrian |
Macro/micro/pico |
|
|
Indoor |
Micro/pico |
|
Figure 11.10: A 3G network supporting mobile, pedestrian, and fixed wireless
Terminal equipment across all of these standards includes multifunction cellular telephones, laptop and tablet PCs, and PDAs and other hand-held computers. Generally speaking, a cellular modem is required for a laptop to establish connectivity, although some manufacturers have started selling computers with built-in wireless antennas. In any case, Internet access via a computer requires a separate subscription agreement that is sensitive to airtime.
11.9.1.4.1 High-Speed Circuit-Switched Data
High-Speed Circuit-Switched Data (HSCSD) is a 2G+ upgrade to GSM designed to improve data transmission rates. Recall that each GSM channel of 200 kHz provides capacity of 270.833 kbps through GMSK modulation. Each channel is divided into eight time slots, which yields a theoretical data rate of 33.8 kbps. Actual throughput, however, generally is limited to 9.6 kbps in consideration of overhead for FEC and encryption. HSCSD improves channel throughput to a maximum of 14.4 kbps in GSM host networks operating at 1800 MHz through the use of improved FEC mechanisms. HSCSD also supports the concatenation (linking) of multiple time slots per frame in support of higher speeds. For example, two time slots yield a transmission rate of up to 28.8 kbps, three yield 43.2 kbps, and four yield 57.6 kbps. In GSM networks operating at 900 MHz, the channel throughput remains 9.6 kbps with concatenation yield maximum throughput of 19.2, 28.8, and 38.4 kbps. As GSM provides circuit-switched, rather than packet-switched, connectivity, HSCSD is more suited for connection-oriented applications such as video and multimedia. E-mail and other bursty data communications applications are served more cost effectively by packet data network protocols running over native packet networks. A small number of carriers have deployed HSCSD as an interim step toward EDGE and UMTS networks. HSCSD has not been deployed in PCS 1900 networks, the upbanded GSM version running at 1900 MHz in the United States [26].
11.9.1.4.2 General Packet Radio Service
General Packet Radio Service (GPRS) is the 2.5G data service enhancement for GSM host networks. GPRS specifications were developed in 1997 by ETSI, which has since passed that responsibility on to the 3rd Generation Partnership Project (3GPP). GPRS is a packet-switched service that takes advantage of available GSM time slots for data communications and supports both X.25 and TCP/IP packet protocols with QoS. GPRS, an important component in the GSM evolution, enables high-speed mobile datacom usage and is considered most useful for bursty data applications such as mobile Internet browsing, e-mail, and various push technologies. Through linking together as many as eight GSM channels, GPRS has a theoretical transmission rate as high as 171.2 kbps, although it realistically is limited to 115.2 kbps and more typically 30–60 kbps. In practice, however, GSM system operators are unlikely to allow a single user to access eight channels. GPRS maintains the same GMSK modulation scheme used by GSM and provides always-on access, although charges apply only for actual data traffic.
Notably, GPRS will support simultaneous voice and data communications over the same wireless link, with voice taking precedence as always. GPRS defines three classes of terminal equipment:
· Class A terminals support simultaneous circuit-switched GSM voice and SMS service as well as GPRS packet-switched data traffic.
· Class B terminals will support nonsimultaneous circuit-switched voice and packet-switched data, automatically switching between the two. A Class B ter-minal, for example, will suspend an active data session in the event of an incoming voice call or SMS message. Most GPRS terminals are Class B.
· Class C terminals support either circuit-switched voice and SMS service or packet-switched services but must be manually switched from one to the other.
The bit rate is sensitive to the encoding scheme in use, of which there are four. CS-4 supports a bit rate of 20.0 kbps per time slot but can be used only when the Mobile Station (MS) and Base Station (BS) are in proximity. As the distance increases between the MS and BS, the encoding scheme must be more robust to compensate for attenuation and MPI, and the bit rate accordingly must adjust downward. At the edge of the cell, for example, CS-1 supports a bit rate of only 8.0kbps. The four Coding Schemes (CSs) and associated bit rates are as follows:
· CS-4: 20.0kbps
· CS-3: 12.0kbps
· CS-2: 14.4kbps
· CS-1: 8.0kbps
GPRS can run in either the symmetric or asymmetric mode, with the speed in either direction sensitive to the multislot service class selected, of which there are 12. The multislot service class determines the number of time slots in each direction, with each time slot supporting a theoretical nominal data rate of 20 kbps (actually 21.4 kbps). The simplest is service class 1, which supports one time slot in each direction. The most capable is service class 12, which supports four time slots in each direction. Generally speaking, the most common service classes are asymmetric in nature, which suits data-oriented Web applications in much the same way as do the asymmetric local loop technologies of ADSL, PON, and WiMAX. The most common GPRS service classes are organized as listed in Table 11.5, which also provides maximum downlink and uplink bit rates based on CS-4 encoding.
|
Table 11.5: GPRS Example Service Classes Open table as spreadsheet |
||||
|
Multislot Service Class |
Downlink Slots |
Uplink Slots |
Uplink Speed [ a ](kbps) |
|
|
2 |
2 |
40 |
1 |
20 |
|
4 |
3 |
60 |
1 |
20 |
|
6 |
3 |
60 |
2 |
40 |
|
8 |
4 |
80 |
1 |
20 |
|
10 |
4 |
80 |
2 |
40 |
|
12 |
4 |
80 |
4 |
80 |
|
[a]Maximum speed, assuming CS-4 encoding. |
The U.S. carriers deploying GPRS include Cingular (800 and 1900 MHz) and T-Mobile (1900 MHz). Upgrades from GPRS to 3G will be in the direction of UMTS, often through EDGE as an intermediate step [26].
11.9.1.4.3 Enhanced Data Rates for GSM Evolution
Enhanced Data rates for GSM (nee Global) Evolution (EDGE) is a 2.5G standard developed by ETSI in 1999 and touted as the final stage in the evolution of data communications within the existing GSM standards. The only IMT-2000 specification based on TDMA, EDGE supports data transmission rates up to 473.6 kbps over GSM FDD channels 200 kHz wide through an improved modulation technique. 8-Phase Shift Keying (8-PSK) involves eight levels of phase shift and, therefore, supports three bits per symbol. EDGE supports 124 FDD channels, each of which supports eight time slots/users. EDGE supports two modes of operation:
· Enhanced GPRS (EGPRS) is a packet-switched transmission mode that will support data rates as high as 473.6 kbps. EGPRS estimates link quality in order to adapt the Modulation and Coding Scheme (MCS), of which there are nine levels, as listed in Table 11.6. If the system estimates that the quality of the link is good, it will select the more efficient 8-PSK modulation technique and, therefore, realize higher signaling rates per time slot and higher data throughput. If the link quality is estimated to be poor, the system will ratchet down to the less capable GMSK. Incremental Redundancy (IR) is an enhanced Automatic Repeat Request (ARQ) technique. As transmission begins, IR initially transmits packets with little FEC overhead in an attempt to maximize efficiency, that is, maximize payload by minimizing overhead. If the initial transmission cannot be successfully decoded by the receiver, IR ratchets up the FEC overhead until it finds a level at which the receiver can successfully decode the transmission.
|
Table 11.6: EGPRS Modulation and Coding Schemes Open table as spreadsheet |
||||
|
MCS |
Modulation Scheme | |||
|
MCS-1 |
8.8 |
143 |
GMSK |
70.4 |
|
MCS-2 |
11.2 |
91 |
GMSK |
89.6 |
|
MCS-3 |
14.8 |
45 |
GMSK |
118.4 |
|
MCS-4 |
17.6 |
22 |
GMSK |
140.8 |
|
MCS-5 |
22.4 |
187 |
8-PSK |
179.2 |
|
MCS-6 |
29.6 |
117 |
8-PSK |
236.8 |
|
MCS-7 |
44.8 |
43 |
8-PSK |
358.4 |
|
MCS-8 |
54.4 |
18 |
8-PSK |
435.2 |
|
MCS-9 |
59.2 |
8 |
8-PSK |
473.6 |
|
Source:[26]. |
||||
|
[b]Overhead as a percentage of payload [b]Signaling speed assuming eight time slots per channel. Throughput = channel capacity-FEC overhead |
· Enhanced Circuit-Switched Data (ECSD) is an enhancement of the native GSM circuit-switched protocol. ECSD adds 8-PSK as a modulation option, thereby increasing the efficiency of data transmission. A GMSK connection requires four time slots to support a 57.6-kbps data rate, but ECSD requires only two.
EDGE also runs over IS-136 TDMA networks in the United States. In either case, EDGE is an intermediate step between 2G TDMA and 3G WCDMA, although some TDMA-based carriers may stop at EDGE. At the time of this writing (June 2007), the Cingular Wireless network (800 and 1900 MHz) largely supports EDGE and the T-Mobile network (1900 MHz) is fully upgraded [26]. Maximum transmission rates typically are in the range of 75–150 kbps.
11.9.1.4.4 Universal Mobile Telecommunications System
Also known as Wideband CDMA (W-CDMA), Universal Mobile Telecommunications System (UMTS) is a 3G technology that is seen as a logical upgrade to GSM, although the two are not compatible. UMTS runs over a carrier 5 MHz wide, compared to the 200-kHz carrier used for narrowband CDMA. UMTS specifications provide for both TDD mode and FDD mode, with TDD largely used in Europe and FDD in the United States. The FDD specifications call for the downlink to run in the 2100-MHz range (2110–2200 MHz) and the uplink in the 1900-MHz range (1885–2025 MHz). As is the case with all true 3G systems, UMTS specifications include 128 kbps for high-mobility applications, 384 kbps for pedestrian speed applications, and 2 Mbps (1.920Mbps) for fixed in-building applications. In reality, UMTS currently caps the transmission rate at a theoretical 384 kbps. UMTS was first deployed in Japan (2000), where it is known as Freedom of Mobile Multimedia Access (FOMA). In some locations in the United States, Cingular has deployed UMTS running in the 850-and 1900-MHz bands. T-Mobile has committed to UMTS worldwide.
UMTS networks currently are being upgraded with High-Speed Downlink Packet Access (HSDPA), which sometimes is characterized as a 3.5G technology. HSDPA promises to increase theoretical downlink data rates to 14.4 Mbps, although current implementations support speeds more typically in the range of 400–700 kbps, bursting up to 3.6 Mbps for short periods of time using an adaptive modulation technique to throttle bit rates up and down as the link permits. HSDPA has been introduced on a limited basis in Austria, Finland, Japan, South Africa, and the United States (Cingular) [26]. Work has begun in the standards bodies on High-Speed Uplink Packet Access (HSUPA). Once increased speeds are in place on both the downlink and uplink, simultaneous voice, data, and even video calls will be quite possible.
11.9.1.4.5 Code Division Multiple Access 2000
Developed by Qualcomm, the company that commercialized CDMA, Code Division Multiple Access 2000 (CDMA2000) is a 3G system based on earlier CDMA versions (also known as TIA/EIA IS-95a and IS-95b). CDMA2000 (also known as IS-856) has been approved by the ITU-R as part of the IMT-2000 family. The initial version, known as CDMA2000 1 × RTT (one times Radio Transmission Technology, with one times referring to standard channel width), offers 2.5G capabilities within a single standard 1.25-MHz channel, effectively doubling the voice capacity of the predecessor 2G cdmaOne systems and offering theoretical data speeds up to 153 kbps (throughput in the range of 70–90 kbps) through the use of QPSK modulation. An enhanced 3G version known as 1 × EV-DO (one carrier EVolution-Data Optimized) is a High-Data-Rate (HDR) version that employs 16-QPSK modulation in support of a peak data rate of 2.4 Mbps on the downlink and 153 kbps on the uplink. 1 × EV-DO supports average aggregate throughput in a fully loaded three-sectored cell of 4.1 Mbps on the downlinks and 660 kbps on the uplinks, with dynamically assigned data rates providing each user with optimum throughput at any given moment. 1 × EV-DO can run in any band (e.g., 450, 800, 1800, and 1900 MHz) and can coexist in any type of network (e.g., CDMA2000, cdmaOne, GSM, TDMA, and AMPS). CDMA2000 runs in the 800-MHz and 1.8–2.0-GHz spectrum. GSM1x is a version designed as a transition specification for GSM operators, involving dual-mode phones. Also known as IS-2000-A, 3x is an enhancement that uses three cdmaOne carriers for total bandwidth of 3.75 MHz. This supports data rates up to 2 Mbps by spreading a multicast signal over the three carriers. In North America, Bell Canada, Sprint Nextel, and Verizon Wireless (800 MHz) have committed to CDMA2000 [26, 34–40]. In August 2006, however, Sprint Nextel announced plans to invest up to US $ 3 billion on an 802.16e, aka mobile WiMAX, network that the company expects to be operational in 2008. Mobile WiMAX is based on Intel technology, rather than Qualcomm's CDMA.
11.9.1.4.6 Time Division Synchronous Code Division Multiple Access
Time Division Synchronous Code Division Multiple Access (TD-SCDMA) is a 3G mobile telecommunications standard being developed in the People's Republic of China (PRC) by the Chinese Academy of Telecommunications Technology (CATT) and adopted by the 3GPP as UTRA TDD 1.28Mcps Option. TD-SCDMA is based on CDMA but uses TDMA as well, with the uplink signal synchronized by the base station. The use of TDMA reduces the number of users contending for each time slot, which has the effect of reducing the technical complexity of the system but at the expense of coverage range and mobility. Like European versions of W-CDMA, TD-SCDMA uses TDD rather than FDD. This approach provides additional flexi-bility in that the uplink and downlink channel capacities can be managed independently. The synchronization of the signal at the base station improves the orthogonality of the coded transmissions, which serves to reduce interference between users and therefore allow increased capacity. TD-SCDMA runs in the 2000-MHz (2-GHz) band, with nominal channel spacing of 1.6 MHz [41]. Asynchronous data rates are highly flexible, ranging from 1.2 kbps to 2 Mbps in both directions. During a field trial in Beijing, it reportedly was possible to make high-quality video calls from a car traveling at 125 km/h at a distance of 21 km from the base station.
While the official reason behind the TD-SCDMA is the avoidance of reliance on Western technology, some suggest that the real reason is the avoidance of patent fees and license fees to Qualcomm and others. Given the fact that there were 190 million users of GSM and CDMA cell phones in the PRC in 2002 and the then current rate of growth was 5 million per month, the PRC certainly had plenty of incentive to avoid those costs. Additionally, it apparently is quite inexpensive to upgrade a cellular network from GSM to TD-SCDMA, while it is very expensive to upgrade a GSM network to UMTS [42]. TD-SCDMA specifications have been finalized, system testing is in progress, and operational systems were expected at the end of 2006.
11.9.1.5 Multimode Cellular
Clearly, there are a lot of cellular standards from which to choose, and carriers have chosen them all in various combinations. Many, but not all, of the 2.5G and 3G standards are rooted in GSM, but the upgrades are by no means automatic and the network of any given carrier at any given location typically supports various combinations of 2G, 2.5G, and 3G. Older mobile stations (i.e., cellular telephones) may be able to access only the 2G network, which is ubiquitous until such time as it is decommissioned, as were the 1G analog networks. Newer mobile stations typically are multimode in nature and, therefore, can access the most advanced and most capable generation of technology available, area by area, for a given carrier. Table 11.7 compares carriers in the United States across cellular technologies. Note that the seemingly odd combinations of technologies that do not follow a natural evolutionary path are due to several factors. First, D-AMPS was essentially an evolutionary dead end. Second, many years of acquisitions and mergers have created kluged networks that eventually will be sorted out as older generations of technology are phased out and eventually decommissioned altogether. In the meantime, the carriers all offer dual-mode and even trimode terminals. They all now offer international multimode sets that are compatible with the original European version of GSM, so that international travelers can communicate abroad.
|
Table 11.7: United States Cellular Standards, by Carrier Open table as spreadsheet |
|||||
|
Generation |
1G |
2G |
2.5G |
3G |
3.5G |
|
Cingular |
AMPS |
|
|
|
|
|
|
|
GSM |
GPRS, EDGE |
UMTS |
HSDPA |
|
|
|
|
|
|
|
|
Sprint |
AMPS |
PCS (CDMA) |
— |
CDMA2000 |
|
|
T-Mobile |
AMPS |
GSM |
GPRS, EDGE |
UMTS |
HSDPA |
|
Verizon |
AMPS |
D-AMPS[a] |
|
|
|
|
|
|
— |
GPRS |
UMTS |
|
|
|
|
PCS (CDMA)[b] |
— |
CDMA2000 |
|
11.9.2 Terminal Equipment
Contemporary cellular terminal equipment takes a number of forms today. Cellular telephones are by far the most common and are available in hundreds of styles, most of which are very small form factor models compared to the early bricks, so called because they literally were about the size and weight of a clay brick. The small form factor certainly has advantages in terms of size and weight but also presents problems with respect to the user interface, as the visual display and data entry keys are so small. These handset issues are of greatest significance with respect to high-end multifunction sets and with data, rather than voice, applications such as Web surfing and SMS messaging.
As the underlying technologies have evolved and 2.5G and 3G mobile stations have become available, cellular telephones increasingly have been oriented toward data, image, and now video applications. Voice, SMS messaging, and text-based e-mail access have been commonplace on cellular telephones since 2000 or so. Instant Messaging (IM) is much more recent. High-resolution color displays combined with high-speed network access have made Web surfing an integral part of the mobile wireless experience for several years. Digital photography and videog-raphy capabilities have now turned the high-end cell phone into an untethered digital camera. In 2004, camera phones made up 36 percent of mobile phone shipments in the United States and more than 50 percent in Europe and dominated the market in Japan [43].
By the way, cell phones do not just ring any longer, they play ring tones, and have done so for some years. Ring tones are not just the tones that you hear when your cell phone rings, they now also can be the tones that the caller hears. Just in case the several dozen ring tones that come with a typical cell phone are not to your liking, you can download additional ring tones for a charge. In fact, ring tones are now a billion dollar business in The United States.
PDAs continue to increase in popularity and ever more are wireless enabled. Similarly, laptop and tablet PCs not only can be interfaced with the cellular networks via PCMCIA cards but also now are available with built-in cellular antennas and software, right out of the box, so to speak.
11.9.3 Present and Future of Cellular Radio
In 1994, Global Telecoms Business estimated that there were approximately 35 million cellular subscribers in more than 150 countries and that cumulative growth rates were more than 30 percent per year. The U.S. market grew by almost 50 percent in 1993 and by 325 percent in Pakistan. It went on to say that wireless accounted for nearly 50 percent of Ericsson's sales, compared to a mere 6 percent in 1984. Lehman Brothers estimated cellular penetration of 9.4 percent by the year 2000, equating to 105 million users of a total accessible population of 1.1 billion [44]. Those estimates turned out to be very conservative. In the United States alone, the FCC estimated that there were 33.786 million cellular subscribers on January 1, 1996 [45]. The Cellular Telecommunications Industry Association (CTIA) estimated that there were over 60 million U.S. users as of June 30, 1998 [46]. Now fast-forward to February 2002, when the ITU-T estimated that there were a billion (1,000,000,000) mobile phones worldwide. Fast-forward again to the end of 2005, at which point the CTIA estimated that there were 207.9 million domestic users at year end 2005, sending SMS messages at the rate of 9.8 billion per month and accounting for a total of 1.6 trillion minutes of airtime for the year [47]. Also as of year end 2005, Informa Tele-coms & Media estimated that wireless subscriptions reached 2 billion (2,000,000,000) worldwide [47]. That means that the number of mobile phones exceeds the number of wired phones. Specifically, there were more mobile phones than wired phones in several European countries and a number of developing countries where wireline infrastructure is noticeably lacking. Perhaps most intriguing is the fact that in Taiwan there are more mobile phones than inhabitants, according to TeleGeography [48]. International Data Corporation (IDC) estimated that worldwide shipment of mobile phones reached approximately 825 million in 2005, an increase of 165 percent over 2004. Nokia, the manufacturer currently with the largest market share, predicts that there will be 3 billion mobile subscribers by 2008 [49]. Without a doubt, cellular radio growth is unprecedented and will continue to grow at phenomenal rates in the foreseeable future.
While Europe and North America may be reaching saturation levels, at least for cellular voice, the developing world is largely an untapped market. The fastest growth rates are in China and Africa. It is estimated that 80 percent of the world's population lives within range of a mobile network but that only about 25 percent have a mobile phone. A good deal of emphasis these days, therefore, is on low-cost cell phones, including prepaid phones, as this combination is especially attractive to low-income users, particularly if they participate in a cash-only economy. As governments increasingly view communications as a basic human right and as a huge stimulant to entrepreneurial economies, they increasingly are lowering taxes, tariffs, and duties on them.
In any event, we certainly are an increasingly mobile society and we clearly insist on remaining in touch. Cellular radio systems satisfy those requirements. And cellular service increasingly is affordable. While average rates were $ 0.45 per minute for airtime in the early 1990s (plus toll charges), they have dropped as low as $ 0.00 per minute in 2002 (with no domestic toll charges), essentially having been flat rated. If you travel as much as I do, you know that this compares very favorably with the cost of a long-distance call from an airport pay phone or hotel room, especially when you add the hotel surcharge. Within the airtime limits of many rate plans, cellular airtime and long distance now essentially are free. Perhaps that explains why, in the United States, approximately 6 percent of households are wireless only, which translates into no wired telephone service [47]. The Yankee Group estimates that since 2003 wireless subscribers in the United States have used their cell phones more than their residential landlines and that, by 2005, personal calling on wireless exceeded that on landlines, even though 35 percent of the population does not have a cell phone [50]. (Note: I have two adult children and Margaret, my lovely bride of 10 years, has one. Each of them has a cell phone, but none of them has a residential landline. I don't quite understand that, but it may be that I'm not supposed to. Emergency calls to 911 services are problematic if the cell phone battery runs down, especially if there has been a power failure and there is no way to recharge it. Emergency services personnel cannot locate the caller with nearly the accuracy they can with a landline. There are lots of other advantages to having a landline, of course.) The trend carries over into the business world as well. In January 2005, Ford Motor Company announced that it was replacing the traditional wireline phones of 8000 employees in its product development department with cellular phones from Sprint [51].
The trend toward cellular radio seems unstoppable, at least for voice and mobile data communications applications. It also appears that the trend will only accelerate as 3G, 3.5G, and even 4G cellular become widely available and data rates increase to truly broadband speeds. As manufacturers and carriers collaborate on multimode devices that can seamlessly bridge the gaps between Wi-Fi, WiMAX, and 3G+ cellular networks, we may very well find ourselves in a largely wireless world. By that time, of course, the definition of broadband will have ratcheted up a number of notches, and wireline networks will deliver multimedia content at Tbps speeds.
In the United States, Sprint Nextel announced in August 2006 that it would spend up to US $ 3 billion to build an 802.16e, aka mobile WiMAX, network slated to begin operation in 2008. The network will be based on technology developed by Intel, which has taken the lead in WiMAX development. Sprint Nextel, with 51.7 million cellular subscribers, expects that the mobile WiMAX network will support connection speeds about five times faster than its current PCS network, which is based on CDMA technology from Qualcomm.
11.9.3.1 Social Implications
The ubiquity of multifunctional cell phones and other mobile devices has clearly changed the way that many people conduct their professional and personal lives. Some years ago, you could be reasonably certain that someone walking down the street waving his arms and yelling at thin air was mentally unbalanced, but today that person is more likely just another loud, rude cell phone user with a Bluetooth-enabled headset. Many restaurants, theaters, and other venues have been declared cell-free zones in defense of these obnoxious people. (If you are one of those people and I have offended you, I sincerely hope that you will get over it—and mend your ways. )
Cellular technology is just an enabler, of course, but it has been put to some uses that are questionable, to say the least. Cell phones with built-in cameras and video recorders have been used to take upskirt or downblouse photos and for various other illicit purposes. As a result, some gyms and health clubs have banned cell phones from locker rooms. In the United States, some states have passed laws against such invasions of privacy. In December 2004, the United States enacted the video Voyeurism Prevention Act, making it illegal to sneak photos or videotapes of people for lascivious purposes on federal land, such as national parks. A study conducted by Juniper Research and quoted in The New York Times indicated that U.S. cellphone adult content was a $ 1 billion business in 2005 and predicted that it would increase to $ 2.1 billion in 2006. Camera phones have been used for good as well. The phones have been used to help catch thieves, burglars, rapists, and all sorts of criminals.
As is the case with many things in life, the greatest benefit of cellular telephony is perhaps also its greatest detriment. The ability to communicate anytime and anyplace can be just as much a curse as a blessing, for it robs us of our personal time. Location-based services can even track us through Global Positioning System (GPS) and Automatic Location Identification (ALI) technologies. These services are handy if you are unfortunate enough to need emergency assistance but can be intrusive if you are unfortunate enough to be bombarded with spim from retailers along your route. Many of us road warriors who travel extensively for business have found airplanes to be a refuge from this constant contact. Unfortunately for those of us, many airlines plan to offer both mobile phone service and Wi-Fi connectivity in the air.
11.9.3.2 Religious Repercussions
As might be expected, religious leaders have taken steps to combat, or at least control, this trend. In 2004, Saudi Arabia's highest religious authority issued an edict banning the use of camera phones, blaming them for spreading obscenity. The concern goes quite far in conservative Muslim societies, where religious authorities fear that camera phones are misused to photograph women without their knowledge and without being properly covered from view, head to toe. According to the Associated Press, a wedding party in July 2004 turned violent after a female guest was caught taking photographs with her phone—scuffles broke out and some guests were hospitalized [52]. (Note: In the previous December, the authorities banned dolls and stuffed animals, so a ban on camera phones should hardly be surprising.)
At about the same time, Mirs Communications, the Israeli subsidiary of Motorola, introduced the kosher phone, which carries the seal of approval of Israel's ultra-Orthodox rabbinical authorities. The kosher phone is stripped down to its bare functional essentials of making and receiving calls. There is no text messaging, no IM, no Internet access, no camera, and no video. Among the more than 10,000 numbers blocked are those for phone sex and dating services. Arab service providers have sought information from Mirs via backchannels using envoys sent from Jordan, which has a peace treaty with Israel [53].
11.9.3.3 Safety Issues
Various safety issues have long swirled around cell phones and their use. Studies conducted over the years have proved and then disproved links between cell phone usage and brain cancer. The most recent of those studies, conducted at the Swedish National Institute for Working Life, indicated that those who heavily used wireless phones had a 240 percent increased risk of a cancer ous tumor on the side of the head where they used their phone. Heavy use was defined as 2000 or more hours, which roughly translates into one hour of usage a day for 10 years. Kjell Mild, who led the study, said, "The way to get the risk down is to use handsfree". Mild did not offer any suggestions with respect to the hands-free technology, although I suspect that Bluetooth might not do a lot to solve the problem. The U.S. Food and Drug Administration plans to review the study [54, 55].
Studies have long shown links between cell phone usage and vehicle accidents. It seems that driving and dialing, and even driving and talking, are distracting and that even cell phones equipped with hands-free options do not eliminate the problem. As a result, some states in the United States have passed laws banning their use while operating a vehicle. As is always the case, some people deny the link and refuse to abide by the law. (Note: They are probably the same people who deny the link between safety and seat belts. You know who you are.)
In June 2006, the British Medical Journal published a letter written by doctors at the Northwick Park Hospital claiming a link between cell phone usage during stormy weather and lightning strikes. They suggest that when someone is struck by lightning, the high resistance of the human skin conducts the flash over the body in what is known as a flashover. If, however, a metal object such as a cell phone is in contact with the skin, it disrupts the flashover and increases the probability of injury and death. The doctors cited three cases reported in China, South Korea, and Malaysia [56].
11.10 PACKET DATA RADIO NETWORKS
Cellular networks of various descriptions have supported data communications for generations, so to speak, but not very well. The 1G analog and 2G digital networks really were optimized for voice, and understandably so. CDPD offered reasonable data communications speeds, but it really was not a satisfactory solution. Alongside those cellular networks low-speed, packet-switched wireless data networks known as Mobitex (Mobile text) were deployed in Europe and the United States, and they remain in operation. Developed jointly by Ericsson and Swedish Telecom, Mobitex offers runs in Europe in the band 400–450 MHz and in the United States in the 800- and 900-MHz bands, although it will run in any band.
Mobitex channel spacing of 12.5 kHz supports theoretical data transmission rates of 8 kbps through GMSK modulation. Mobitex packets are up to 512 octets, including overhead, using a slotted Aloha protocol. Slotted Aloha is an improvement on the original Aloha protocol developed at the University of Hawaii in the early 1970s as a contention management mechanism for use in interisland wireless networks. The basic concept behind the Aloha protocol later was incorporated into the Carrier Sense Multiple Access (CSMA) protocols that Bob Metcalf and his associates developed for Ethernet.
Mobitex networks support SMS messaging, corporate e-mail access, and Internet access. Web surfing is supported through proprietary gateways that can filter out graphics or lower their resolution to reduce their bandwidth intensity. Security is provided through various proprietary authentication and encryption mechanisms. Applications include sales agents, maintenance fleets, and truck fleets. Terminal equipment supported includes two-way pagers, PDAs and various other hand helds, and laptops. Mobitex networks largely were retired in Europe in the face of strong competition from GSM but remain in place in 30 countries, including in the United States and Canada. These networks face strong competition from 2.5G and 3G cellular networks that support voice, SMS and MMS, e-mail, and Internet access from traditional cellular phones, laptop and tablet PCs, and PDAs and other hand helds.
11.11 SATELLITE SYSTEMS: LEOs, MEOs, AND GEOs
I discuss satellite systems in quite some detail in Chapter 2, where we explored the basic technology with a focus on GEO (Geosynchronous Earth-Orbiting) systems for point-to-multipoint, or broadcast, applications. As the focus of this chapter is on mobility, I examine satellite systems that support mobile voice and data applications.
Low-Earth Orbiting and Middle-Earth Orbiting satellites (LEO and MEO) satellites operate at low altitudes of several hundred miles or so in a variety of nonequa-torial orbital planes. This compares with GEO satellites, which always are placed in equatorial orbital slots at an altitude of approximately 22,300 miles. LEO satellites operate at altitudes of 644–2415 km. Although the term is not tightly defined, little LEO systems involve a relatively small number of satellites and operate at frequencies below 1 GHz in support of low-bit-rate data traffic (e.g., telemetry, vehicle messaging, and personal messaging). Big LEO systems are bigger networks that operate at higher frequencies in support of voice and higher speed data communications. MEO satellites operate at altitudes of 10,062–20,940 km.
LEO and MEO systems are configured as constellations of small, low-power satellites. In combination, the satellites in such a constellation generally provide full coverage of major land masses, and some have been designed to provide full coverage of virtually every square inch of the earth's surface. The various proposals have included as many as 840 satellites and are intended to provide various combinations of voice and data services. These systems also are known as Mobile Satellite Systems (MSSs), as opposed to the Fixed Satellite Systems (FSSs) in geostatic orbit. Whizzing around the earth like electrons whizzing around the nucleus of an atom and as illustrated in Figure 11.11, LEO and MEO networks are designed so that a satellite is always within reach of a terrestrial terminal.
Figure 11.11: LEO satellite constellation
11.11.1 How LEOs Work: Iridium
The origin of this incredible concept is worth exploring. According to legend, the wife of a Motorola executive was vacationing in the Bahamas during 1987 and was irritated by her inability to place a cellular telephone call (probably from some secluded beach not covered during those early days of cellular service). She complained to her husband and captured his imagination. Motorola named its proposed 77-satellite constellation Iridium, after the element Iridium (Ir), which boasts 77 electrons whizzing around its nucleus. Iridium, by the way, gets its name from the Latin and Greek irid and iris, meaning rainbow, after the rainbow colors of the metal when dissolved in hydrochloric acid. (Iris was the Greek goddess of the rainbow and a messenger of the gods.) Subsequently, the proposal was pared down to 66 operational satellites, although the name Iridium stuck. After all, Iridium seems to roll off the tongue better than Dysprosium (Dy), the rare earth element with an atomic weight of 66. Dysprosium, by the way is from the Greek dysprositos, meaning hard to get at, which is hardly a good name for a communications technology. The Iridium constellation is now fully launched and is fully operational. Eleven operational satellites and one spare are placed in each of the six orbital planes at altitudes of 421.5 nautical miles. Connectivity between each satellite and the earth is established via 48 highly focused spot beams, each of which has a footprint of approximately 30 miles (50 km).
The functioning of the Iridium network, which is the most complex operational constellation at the present, will serve as an excellent example of LEO systems, in general. As illustrated in Figure 11.12, you might initiate a voice call from one Iridium phone to another Iridium phone using a special international telephone number in a block allocated to Iridium. The telephones are slightly supersized versions of cellular phones. Since the frequency band is in the gigahertz range, Line-of-Sight (LoS) is critical, so you either need to be out in the open or you need to stick a little satellite dish on top of your automobile. Your Iridium phone, by the way, is equipped with an antenna the size of a good cigar, which you first must unfold. Your connection directly to the satellite is supported over the L-band at frequencies of 1.616–1.6265 GHz. If you connect from a landline through a local terrestrial gateway, Ka-band frequencies are used, with 29.1–29.5 GHz on the uplink and 19.3–19.7 GHz on the downlink. Once connected to the satellite, the intersatel-lite links operate in the Ka-band at frequencies of 23.18–23.38 GHz. (See Table 2.5 for a listing of satellite frequency bands.) The satellite constellation finds the target Iridium phone just like a cellular network finds another cell phone. Each active (powered-up) phone maintains a signaling and control link with the satellite network, just as your cell phone keeps in touch with the cellular network when it powers-up. Presto, you're connected, and the quality resembles that of an analog AMPS network, which is pretty good for a satellite call.
Propagation delay is not much of an issue for several reasons. First, the satellites are in low-earth orbits of approximately 485 miles, so the uplink and downlink propagation delays are not significant. Second, the satellites communicate directly over switched intersatellite links, so only one uplink and one downlink are required. As the satellites whiz around the earth like electrons whiz around the nucleus of an atom, however, they have a very short dwell time, so you cannot maintain contact with any given satellite for very long, and neither can the other party involved in this telephone call. Therefore, the satellite with which you established the connection must pass that call off to another satellite before it gets out of view, and so must the satellite at the terminating end, and so must every satellite in between. Think of it as a switched cellular network in reverse. In a cellular network, the antennas at the cell sites are stationary. As you whiz through the cellular network in your high-speed vehicle, the cellular network maintains the connection through a hand-off process between cell sites. In an Iridium environment, you are the one who is (relatively) stationary, while the cell sites whiz around you. Iridium also, by the way, supports pagers and airplane communications via the same L-band frequencies. If you call a device that is not on the Iridium network, your call can be connected to the existing PSTN and cellular networks via 12 regional gateways.
Iridium is an incredible network, and it should be, as the total capital investment was in the neighborhood of $ 4.7 billion. The system went fully operational on November 1, 1998. On May 28, 1999, Iridium LLC received a waiver until June 30 of certain financial covenants from all its lenders to enable it to restructure its capitalization. Those covenants required that the company have at least 27,000 customers by May 31. It seems as though the actual numbers fell far short of that requirement. At a cost of $ 3000 or so for an Iridium phone, $ 3795 for a dual-mode phone that also works on existing cellular networks, $ 500 or so for a pager (one-way, so no guaranteed message delivery), some modest amount of money for a cable and data modem, a considerable monthly charge, and $ 3–$ 7 a minute for the connect time (depending on which press releases and articles you believe), Iridium was pretty pricey. Most commercial users just could not justify the cost unless they wanted to call between very remote areas such as the summit of Mt. Everest (true story) to the jungles of New Guinea (it is possible if the canopy of the rain forest is not too thick) [57–67]. Iridium's high cost and relatively poor service did not sit well with prospective users, who failed to appear in numbers even close to those required to make the network financially viable. Iridium filed for Chapter 11 bankruptcy in August 1999 and ceased service in April 2000. At that point, the plan was to let the satellites naturally degrade in their orbits until they simply burned up in the atmosphere in a spectacular Iridium flambé (at least that's the way I like to think of it). Rather than let Iridium flame out, however, a group of investors purchased the network for a paltry $ 25 million. Shortly thereafter, in what can only be termed an interesting coincidence, the U.S. Department of Defense agreed to a $ 72 million, two-year contract to provide service to 20,000 government users. Other users can subscribe to Iridium services through one of the 19 or so authorized service partners in various countries around the world. Costs for airtime have dropped to a range of $ 1.40–$ 3.00 per minute, and costs for handsets have dropped to $ 1500 or less. Analogies such as Icarus and Phoenix aside, suffice it to say that Iridium's close call with a fiery death put a real damper on a number of other proposed LEO projects, and Iridium's long-term financial viability remains in question, to put it mildly. After all, the constellation must be refreshed periodically to avoid future flambé.
By the way, there is a lesson to be learned through Iridium, at least in my humble opinion. The Iridium concept was a great one in 1987 and even through the early 1990s. At that time, the cellular networks were not fully built out and its level of acceptance was not envisioned as ever reaching the current levels, DSL and CATV modems did not exist, and WiMAX and PON were not even gleams in engineer's eyes. Somewhere along the way, however, the folks at Motorola just got too committed to Iridium, even as the competing technologies developed. So, they and others continued to pump billions of dollars into a clever idea that never had a chance. They just breathed their own exhaust (i.e., believed their own marketing hype) until they became so intoxicated that they lost touch with reality. Hindsight is always perfect, but even a little foresight in the mid-1990s would have made it quite clear that Iridium was doomed. Having said that, I also must say that all reliable reports suggest that Motorola made plenty of money off of the idea—at the expense of the other partners. So, maybe Motorola had it right after all, even though other investors lost billions.
11.11.2 How LEOs Don't Work: Teledesic
Let us now turn our attention to the Teledesic LEO network, dubbed the Internet-in-the-Sky. Teledesic is a combination of tele, meaning over a distance, and geodesic, meaning the shortest distance between two points that lie on a given surface (i.e., the earth). Teledesic was the brainchild of Bill Gates of Microsoft fame, Craig McCaw of McCaw Cellular (acquired by AT&T Wireless, which then was acquired by Cingular) fame, and Boeing of Boeing aerospace fame. Gates knows a lot about computers and the Internet, McCaw knows a lot about wireless, and Boeing knows a lot about satellites and launch vehicles. Motorola also joined this group at one time. Investors included Bill Gates, Craig McCaw, His Royal Highness Prince Alwaleed Bin Talal Bin Abdul Aziz Alsaud (Saudi Arabia), Abu Dhabi Investment Company, and The Boeing Company. (By the way, they all know a lot about making money.) The idea behind Teledesic was to launch a constellation of 30 (pared down from 288, which was pared down from the original 840) broadband LEOs, plus spares, for Internet access. These two-way satellites were intended to support high-speed Internet access from the subscribers' premises in the Ka-band at rates from 128 kbps to 100 Mbps on the uplink (28.6–29.1 GHz using FDMA) and up to 720 Mbps on the downlink (18.8–19.3 GHz using TDMA). The satellites would whiz around your inexpensive, two-way satellite antenna at low altitudes (about 700 km, or 434 miles), thereby avoiding the aggravating issues of propagation delay that might otherwise have you yelling at your computer. From your computer, you would have a connection to your dish, to the satellite, from satellite to satellite, and down to a portal, which would be your gateway to the Internet and the Web. The costs of your access to this multi-billion-dollar network were expected to be competitive when it would become operational in 2005 or so [57, 58, 62–65]. That was the basic idea, and it was an incredible one—if the technology had fallen into place—and there certainly was no technical reason that it could not. There were a lot of very smart people behind this project, with very deep pockets playing for very big stakes. If you stop to think about Teledesic for just a minute, you will realize that it would have bypassed your modem, your local loop, your Local Exchange Carrier (LEC), your Internet Service Provider (ISP), the wired Internet backbone and all its pieceparts, and your current portal. Teledesic would have bypassed everything conventional. (Thinking back to the definition of Teledesic at the beginning of this discussion, you now understand that the term means that the shortest distance between your computer and the Teledesic portal is the Teledesic network, which seems to have been a codeword for Microsoft Network, or MSN.)
Now, I may not be as smart as the Teledesic folks, and I certainly do not have pockets nearly as deep as the least rich of the lot, but I bet my next royalty check from this book that the Teledesic portal will not be optimized for Netscape Navigator. As I said before, it is an incredible idea. Note, however, that Teledesic was conceived in 1986, when there were no optical fiber backbones, no DSL, no cable modems, no PON, and no WiMAX and when Internet access typically involved modems that ran at the blazing speed of 19.2 kbps, best case. In October 2002, Tele-desic suspended construction work, and in July 2003, the company gave up its frequency licenses. So, another great idea conceived by engineers went kerplooey.
In the meantime, Craig McCaw and a subset of the principals of the failed Tele-desic venture had moved on to ICO, for Intermediate Circular Orbit, which they rescued from bankruptcy in 2000. The company reemerged as New ICO, and the system was redesigned. New ICO is a MEO design slated for 10 operational satellites in two orbital planes. After a failed launch or two, New ICO faded into the background for several years. As recently as April 2005, however, New ICO announced that Space Systems/Loral was awarded a contract to build a geostationary mobile satellite system. So, it appears that the system was redesigned yet again and is now a GEO rather a MEO design.
11.11.3 GEOs Work Too: Inmarsat
Inmarsat, for International Maritime Sat ellite, is the pioneer in the MSS space, having been in service since the early 1990s. Inmarsat also is based on a bent-pipe GEO design, with no intersatellite links. Each Inmarsat GEO generates 19 wide spot beams and 228 narrow spot beams. The system reportedly covers 98 percent of the world's population with services including voice, e-mail, Internet access, fax, and videoconferencing at speeds up to 492 kbps [66].
11.12 AND THAT'S NOT ALL
The creativity and inventiveness of the modern human mind should not be underestimated, for there is still more technology to explore, although some of it may seem a bit odd, if not downright wacky. Remember, however, that the telephone seemed like a wacky idea to many people just 130 years ago. Please read on.
Platforms Wireless International plans to develop an unmanned helium-filled aerostat (i.e., static blimp) that will provide inexpensive wireless communications access to rural areas around the world. The 1250-lb antenna payload is mounted on an aerostat roughly twice the size of the Goodyear blimp, tethered to a ground mooring station by a Kevlar cable and floating at an altitude of approximately 3500–4500m in restricted airspace. The platform technology is well established, as the U.S. Border Patrol uses similar aerostats for surveillance along the Mexican border. The ARC (Airborne Relay Communications) system can provide cellular coverage to an area roughly 140 miles in diameter. In the Americas, the system will use TDMA cellular technology, thereby avoiding the issue of expensive terminal equipment. Systems installed elsewhere during subsequent phases will use GSM or other appropriate cellular standards. The initial implementation target is in Brazil, where telephone penetration is very low [67–69].
Along similar lines, Sanswire Networks plans to offer a variety of wireless services via high-altitude airships known as Stratellites. The airships will take up positions in the stratosphere at 65,000 ft (approximately 13 miles, or 20 km) and hold to a single GPS coordinate through the use of solar-powered electrical engines. Each Stratellite will support a payload of several thousand pounds and have clear Line-Of-Sight (LOS) coverage of about 300,000 square miles, an area roughly the size of Texas, according to company information [70].
And there is more. Angel Technologies Corporation plans to develop a high-capacity wireless metropolitan network dubbed HALO (High-Altitude Long Operation). The network will use shifts of specially designed aircraft capable of carrying a telecommunications payload of 2300 lb and remaining aloft for 8h or so at an altitude of approximately 51,000 ft. At that altitude, the coverage area is about 75 miles in diameter. HALO is expected to provide raw aggregate capacity exceeding 16 Gbps, in support of 50,000–100,000 customers requiring symmetric T1 packet data rates [71].
Chapter 12: Video and Multimedia Networking
OVERVIEW
"It's a success, Ned! I've struck it!" cried Tom, in delight.
"Ouch! You struck ‘me,’ you mean!" replied Ned, rubbing his shoulder, where the young inventor had imparted a resounding blow of joy.
"What of it?" exclaimed Tom. "My apparatus works! I can send a picture by telephone! It's great, Ned!"
"But I don't exactly understand how it happened," said Ned, in some bewilderment, as he gazed at the selenium plate.
"Neither do I," admitted Tom.
Victor Appleton, Tom Swift and His Photo Telephone or The Picture That Saved a Fortune, Gross & Dunlap, 1914
Tom Swift certainly had a great idea, although he did not understand quite how his photo telephone worked. Tom used his invention to save Mr. Damon from Messrs. Peters and Boyman, kidnappers and swindlers who were close to cheating him out of a fortune. While the dastardly pair had poor Mr. Damon kidnapped, they called Tom to arrange for the ransom. Tom took a picture of them with his photo phone. That photo ultimately was used as evidence to convict them. While Tom may have saved Mr. Damon a fortune, there is no mention of the cost of the photo phone. However, I must assume that the cost was reasonable, as the young inventor was featured in an entire series of books in which he also invented the motorcycle, the motorboat, the air glider, the airship, the submarine, the wireless telegraph, the electric rifle, and a giant cannon—all funded with his modest allowance. Since Tom invented the photo phone, the concept has been refined by other young inventors and now is used in a variety of applications, including law enforcement. In the right applications, the technology clearly has the potential to save a fortune, but the costs tend to be more than typical teenagers can afford out of their allowances. After all, Tom Swift's invention was just a simple image transmission over an analog telephone circuit. Alexander Graham Bell's photophone, on the other hand, was voice only, certainly was not funded out of his allowance, and there is no evidence that it was used to solve any crimes. This chapter deals primarily with interactive video and multimedia networking, neither of which Alexander Graham Bell or Tom Swift could have imagined—or could they?
On the whole, you may wonder how critical interactive video and multimedia networking are to our daily lives. The answer is not clear. While we all understand that a picture can be worth a thousand words and we clearly benefit from one-way video and multimedia, the ability to engage in an interactive video or multimedia communications is not necessarily that important to most of us, either personally or professionally. Such interactive networking currently is expensive in terms of both bandwidth and equipment, although those costs certainly will come down over time. Many of the highly touted benefits have yet to be proven, although a compelling case is developing.
Certainly, video and multimedia have application in education. People tend to have different learning styles. Some are visual learners and need to see something in order to process the information. Some are auditory and need to hear about it. Still others tend to be more kinesthetic and need to touch it. In any event, the addition of the visual information stream certainly assists in the educational process, as it does in the communications process, in general. A fascinating application is in the world of health care. As discussed later in this chapter, telemedicine has proved itself to be of great importance in the delivery of high-quality medical care to remote areas where specialists, and even general practitioners, may not be available.
12.1 VIDEO COMMUNICATIONS: DEFINED AND EVOLVED
The first public demonstration by wire and wireless of Television or "Distant Seeing," as developed by the staff of the Bell System, took place on April 7, 1927. Participating in the demonstration at Washington, D.C., and New York were notable gatherings of leaders in the fields of science, industry and public affairs. Those who talked from the Bell Laboratories in New York were able to see plainly the features of those in Washington with whom they conversed over the long distance circuits of the Bell System. [This] … was followed by demonstration of Television by radio in which the audience in New York saw the artists visualized on the screen and heard a varied program from the radio experimental station of the Laboratories at Whippany, N.J.
Things Worth Knowing about the Telephone, American Telephone & Telegraph Company, 1931
Video (from the Latin video, meaning I see) communications has its roots in broadcast TeleVision (TV), from the Greek tele, meaning far off, and the Latin vision, meaning to see. Broadcast TV is transmitted over the airwaves and remains the primary source of TV in many areas of the United States and in most other countries. The first true TV mechanism was developed in 1884 by Paul Nipkow, a German engineer, using a scanning disk, lenses, mirrors, a selenium cell (as did Tom Swift), and electrical conductors. Doing so, he was able to transmit images in rapid succession to a lamp, which changed in brightness according to the strength of the currents received. Using this mechanical scanning technique, Nipkow demonstrated that portions of a full image viewed in rapid succession (15 images or more per second) created the illusion of viewing the full image [1]. It later was discovered that viewing 15 or more images per second created the illusion of full motion, due to electrochemical processing delays in the human eye. While this mechanical scanning approach was abandoned in later years, the concept of persistence of vision remains valid. Psychologist and film theorist Hugo Munsterberg explained a second process, known as the phi phenomenon, in 1916. This process explains the fact that we hallucinate, or believe that we see, a continuous action rather than a series of still images—the mind, in effect, fills in the blanks [2]. Modern TV and video systems create the illusion of motion by refreshing screens in rapid succession, a dot at a time and a line at a time. (Note: Film projectors refresh the entire screen at once.)
The development of modern TV largely was due to the efforts of Herbert E. Ives, a scientist at Bell Telephone Laboratories. As the son of Frederick Eugene Ives, who in 1878 developed the first practical process for making halftone printing plates, Ives was oriented toward the visual world. In 1923, he and his associates combined the photoelectric cell with the vacuum tube repeater to produce the first commercial system for the rapid transmission of pictures over telephone wires for application by the daily press. Using a scanning beam developed by Frank Gray, another Bell Labs scientist, multiple real-time images of people were transmitted in rapid succession, and TV was born. The first public demonstration, on a black-and-white TV, was conducted in April 1927; color TV first was demonstrated in June 1929 [3]. That experimental color transmission included the transmission of pictures of an American flag, a watermelon, and a bunch of roses. The transmitting system had three sets of photoelectric cells, amplifiers, and glow-tubes, with each filtering out one color—Red, Green, or Blue (RGB). At the receiving end, mirrors superimposed the monochromatic images to create a single color image [1]. In the 1930s, the first commercial TV stations began operation over the radio waves.
Coaxial cable entered the world of TV in 1936, when the first experimental transmission took place between New York and Philadelphia. Jointly conducted by AT&T and the Philadelphia Electric Storage Battery Company (Philco), the experi-ment proved highly successful in terms of transmission performance, as multiple frequencies could be transmitted over the same shielded medium. In 1950, AT&T opened the first coaxial cable for coast-to-coast TV transmission [4]. Community Antenna TeleVision (CATV), generally based on coaxial cable, largely has sup-planted broadcast TV in the United States. The popularity of cable is due in large part to the large number and wide variety of available channels; broadcast TV simply cannot compete at this level due to spectrum limitations. More recently, CATV spread to a number of other developed countries, including England and Australia.
Now, only two key evolutionary concepts remain—those of the transmitters and receivers. Vladimir Zworykin, a Russian immigrant to the United States, built on his graduate work in Russia where he had studied the nature of fluorescence under Boris Rosing. On January 1, 1939, Zworykin received patents for his iconoscope (transmitting) and kinescope (receiving) tubes, which formed the basis for modern cameras as well as Cathode Ray Tubes (CRTs)used as display devices in traditional TV sets and computer monitors [1, 5].
Video communications extends well beyond broadcast TV and CATV into videoconferencing, multimedia communications, and, ultimately, interactive TV. In this chapter, I explore the nature of the equipment and networks that support video and multimedia collaborative communications as well as related standards and costs.
12.2 VIDEO BASICS
In order to fully comprehend the nature and implications of advanced video communications, it is necessary to understand the basics of video. The basic concepts are frame rate, scanning, resolution, aspect ratio, luminance, chrominance, and synchronization.
· Frame rate refers to the rate at which frames of still images are transmitted. Video is a series of still images that are transmitted in succession to create the perception of fluidity of motion. If transmitted in rapid succession, the perception is one of complete fluidity; 24 frames per second (fps) is considered to be motion picture quality, and 30 fps is considered to be broadcast quality. If the frames are transmitted at a slow rate, the result is a poor-quality, herky-jerky video that creates a strobe-light effect reminiscent of 1970s discos. Particularly below 15 fps, quality suffers quite noticeably, as the fluidity of motion is lost even though the image quality may be quite high.
· Scanning refers to the process of refreshing the screen; the scanning rate is a function of the power source of the receiver. Interlaced scanning, which is used with most analog TV systems, involves two fields. Odd lines (field 1) are refreshed in one scan and even lines (field 2) in the next. Each set of odd and even lines refreshed constitutes a frame refreshed. For example, the American NTSC standard provides for 30 fps, involving 60 scans, which relates directly to the 60 Hz of the U.S. power source. The European PAL standard provides for 25 fps, involving 50 scans, which relates directly to the 50 Hz of the European power source. Progressive scanning involves displaying all horizontal scan lines in one frame at the same time, which avoids the problem of interline flicker. While the refresh rate varies, most contemporary PC monitors operate at 60-90 fps. The difference between the two techniques is imperceptible to most of us, until we see a broadcast TV news show that includes a video clip of PC monitors.
· Resolution refers to the definition, or sharpness, of the image. Resolution is determined by the number and areal density of the pixels, or pels (picture elements), which essentially are dots of picture similar to the dots in half-tone printing. The greater the number and density of the pixels, the better the resolution. If the same number of pixels is spread over a greater area, the result is a grainier picture, as you can readily see by sitting close to a big-screen TV.
· Aspect ratio refers to the relationship between the width and the height of the image. The 4: 3 (4 wide to 3 high) aspect ratio specified by the American NTSC standard is rooted in the early days of TV, when round picture tubes made effective use of this approach.
· Luminance refers to intensity, or brightness, which can vary within an image. An analog video transmission varies the luminance by varying the power level, or amplitude, of the signal, with high power representing black and low power representing white. (Note: Broadcast TV from distant stations often displays as grey due to signal attenuation.)
· Chrominance refers to color, with different standards permitting varying levels of color depth. Clearly, the video image is more pleasing and lifelike when the range of color is as broad as possible.
· Synchronization between the transmitter and receiver includes vertical and horizontal synch, both of which are critical. Vertical synch is required to keep the picture from scrolling, or flipping. Horizontal synch keeps the picture from twisting.
12.3 ANALOG TV STANDARDS
Television standards are several and incompatible. The initial standards were set in the United States, where broadcast TV originated. In 1945, the Federal Communications Commission (FCC) set the initial VHF (Very High Frequency) transmission standards at 4.5 MHz. The National Television Standards Committee (NTSC) was formed in 1948 to standardize the characteristics of the broadcast signal. Ultimately, the Radio Corporation of America (RCA), which was owned by AT&T, lobbied the Electronics Institute of America (EIA) and set the initial black-and-white TV standards. Color TV was commercialized some years later. Among the first live color TV broadcasts were the Cotton Bowl and Rose Bowl football games on January 1, 1954 [1]. As the cost of color TV sets was quite high, color television did not really become popular until the 1960s when sets became affordable for the masses. Table 12.1 compares the major analog standards of NTSC and PAL, which are discussed in more detail in the bulleted list that follows, with digital HDTV.
· National Television Standards Committee (NTSC) was established in the United States as the first standard (1953), setting the tone for broadcast TV. NTSC is defined in ITU-R Recommendation 1125. While other standards have since been developed, they all derived from the NTSC baseline. NTSC is characterized as analog in nature, with 525 interlaced scanlines. There are 640 pixels per line, 485 of which are dedicated to the active picture. The frame rate is 30 fps, 60 fields interlaced, and the aspect ratio is 4: 3. An early analog standard that is viewed as overly complex and ineffective in the contemporary digital world, NTSC also is said to mean Never The Same Color [6].
· Phase Alternate Line (PAL) was established in West Germany, The Netherlands, and the United Kingdom in 1967. PAL addresses problems of uneven color reproduction that plague NTSC due to phase errors associated with electromagnetic signal propagation. PAL inverts the color signal by 180 ° on alternate lines, hence the term phase alternate line [6]; it currently is used in much of Western Europe, Australia, and Africa. PAL is characterized as analog, with 625 interlaced scanlines. There are 640 pixels per line, with 576 dedicated to the active picture. The frame rate is 25 fps, and the aspect ratio is 4: 3.
· Séquential Couleur Avec Mémoire (SECAM) (which translates as sequential color with memory), a variation of PAL, was developed in France. In addition to its use in France, it also is the standard in regions once under French influ-ence, including areas of the Middle East.
|
Table 12.1: Television Standards Compared Open table as spreadsheet |
|||
|
Standard |
NTSC |
PAL |
HDTV |
|
Analog/digital |
Analog |
Analog |
Digital |
|
Horizontal scanlines |
525 |
625 |
640, 704, 1280, or 1920 |
|
Synchronization |
40 |
49 |
N/A |
|
Resolution, pixels per line |
640 |
640 |
480, 720, or 1080 |
|
30i |
25i |
24p, 30p, 60p, and 60i |
|
|
Aspect ratio |
4:3 |
4:3 |
4:3 and 16:9 |
|
[a]i = interlaced, p = progressive. |
12.4 DIGITAL TV AND HIGH-DEFINITION TV
Digital TV (DTV) transmission offers the same advantages as any other form of digital communications, including enhanced bandwidth efficiency through compression, improved signal quality, and more effective management and control. Additionally, digital video content must be recognized as nothing more than video data. As a result, it offers advantages in terms of processing, storage, and manipulation. Those advantages include editing, alteration (e.g., morphing), reproduction, compression, and store-and-forward capability.
TV and most video, for that matter, in their native forms comprise synchronized analog image and analog voice information. Digital video requires that the information be digitized through the use of a video codec. A broadcast-quality video signal is extremely bandwidth and storage intensive. For example, the resolution of a digital video might require 640 horizontal and 480 vertical pixels. As chrominance and luminance require 24 bits per pixel, the video signal, alone, requires 7,372,800 bits per frame. At a frame rate of 30 fps, the bandwidth requirement is 221,184,000 bps. Clearly, it is not within the realm of reason to store and transmit a broadcast-quality video signal requiring 221+ Mbps. It is clear, therefore, that compression is critical if the networks are to support digital TV.
Despite all the advantages of digital technology as applied to TV production, storage, transmission, and reception, all of the TV sets were analog. A conversion from analog to digital broadcast TV is revolutionary and completely unthinkable, unless there exists a compelling reason for the changeout of untold millions of TV sets. According to the manufacturers and the FCC, that compelling reason is in the form of High-Definition TV (HDTV).
The definition of HDTV standards caused a debate that raged for a number of years between Japanese analog standards and digital standards proposed by the United States. The U.S. digital standards proposals focused on those offered by the Grand Alliance, which was formed by the FCC in May 1993 and comprised AT&T, General Instruments, Zenith, The Massachusetts Institute of Technology (MIT), Thompson Consumer Electronics, Philips Consumer Electronics, and the David Sarnoff Research Center. The efforts of the Grand Alliance led to a set of recommended Digital TV (DTV) standards for HDTV and Standard-Definition TV (SDTV). Those recommended standards were tested and documented by the Advanced Television Systems Committee (ATSC) in the summer of 1995 and approved by the FCC in December 1996. Table 12.2 presents the specifics of the ATSC scanning formats.
|
Table 12.2: Scanning Formats: ATSC Digital Television Standards Open table as spreadsheet |
||||
|
Standard |
Vertical Lines |
Horizontal Pixels |
Aspect Ratio | |
|
HDTV |
1080 |
1920 |
16:9 |
24p, 30p, 60i |
|
|
720 |
1280 |
16:9 |
24p, 30p, 60p |
|
SDTV |
480 |
704 |
16:9,4:3 |
24p,30p,60p, and 60i |
|
|
480 |
640 |
4:3 |
24p, 30p, 60p, and 60i |
|
Source: [7, 8]. |
||||
|
[a]i = interlaced, p = progressive. |
Considering that an HDTV video signal at 1080 × 1920 pixels, 24 bits per pixel, and 30 frames per second requires bandwidth of almost 1.5 Gbps, compression becomes very important. The ATSC standard specifies the compression algorithm as MPEG-2, which is discussed later in this chapter and the transport subsystem as ISO/IEC 13818. Packet transport involves a serial data stream of packets of 188 octets, 1 octet of which is a synchronization byte and 187 octets of which are payload. This packet approach is suitable for Asynchronous Transfer Mode (ATM) switching, as each 188-octet MPEG-2 packet fits nicely into the payload of four ATM cells, with only 4 octets of padding required. Forward Error Correction (FEC) is employed in the form of Reed–Solomon coding. Radio Frequency (RF) modulation is accomplished using 8 Vestigial Sideband (8 VSB), which supports a bit rate of 19.28 Mbps over a 6-MHz terrestrial broadcast channel through the use of eight discrete amplitude levels. Audio compression is based on the AC-3 specification from Dolby Digital and the ATSC. The audio sampling rate is 48 kHz, and the system supports six channels in the Dolby Digital surround format. That format specifies multiple channel outputs, including center, left and right center, left and right surround, and Low-Frequency Enhancement (LFE), also known as subwoofer [7]. Despite the fact that most programming is not yet in HDTV format and that most broadcast and cable signals are still analog, Forrester Research estimates that about 15.7 million households had HDTV sets by the end of 2005 and predicts that number to be 50 million by 2009 [9].
The FCC set a timetable for DTV transmission, then changed it several times and softened the requirements after the majority of broadcasters missed the first deadlines. Beginning on November 1, 1998, 42 stations (including noncommercial stations) in the top − 10 markets began voluntary DTV transmissions, in addition to analog NTSC transmissions. All stations in those markets were required to transmit DTV by May 1999. After that deadline was missed by a number of broadcasters, the FCC deferred, and then deferred again, its requirement that all commercial TV stations convert fully to digital and fully replicate their analog broadcast signals or lose interference protection. That date ultimately was changed to July 1, 2006, with provisions for waivers. In the meantime, broadcasters constructed and operated lower powered DTV facilities to at least their community of license, extending them to their entire service area as the transition progressed. By February 17, 2009, all stations must turn off their analog signals. While the FCC made the DTV RF spectrum available at no cost to the broadcasters, each station is responsible for any costs associated with clearing DTV spectrum of other users. The analog spectrum will be returned for reassignment at the point in time at which 85 percent of the television households in a given market are capable of receiving the digital channels.
Each station also is responsible for the costs associated with upgrades to digital transmission facilities. The costs of converting the several hundred million analog TV receivers are the responsibility of the viewers, although beginning in 2008, the National Telecommunications and Information Administration (NTIA) will provide to eligible households up to two $ 40 vouchers good toward the purchase of digital-to-analog converters to retrofit existing analog TV sets, Video Cassette Recorders (VCRs), and other devices. New TV sets, with screens from 25 to 36 in., were required to have built-in digital tuners as of March 1, 2006. By March 2007, all new TVs, regardless of size, and other devices designed to receive broadcast TV signals must have digital tuners built in. According to U.S. census projections, there were approximately 110 million households in the United States at the end of 2005 [10, 11]. Considering the fact that the average household had 2.6 television sets, the costs of adding 286 million analog TV sets and 100 million or so VCRs to existing landfills will be substantial. Assuming the use of a single landfill, just imagine its size. Just imagine the bewilderment of future archeologists when they excavate it. (This assumes that there will be future generations, which is questionable, since the mercury and lead that leaches out of the TV sets and out of the landfill into the streams, rivers, lakes, and oceans of the world may have killed off all life as we know it.) They will probably think it was some sort of sacrifice to the TV gods, and they will not be far off. In any event and as a byproduct of the conversion from analog to digital broadcast TV, the 750 MHz of analog spectrum will be reclaimed for fixed and mobile commercial wireless services, and that spectrum will be auctioned once the reclamation is complete [12, 13].
You may wonder why I spent so much time discussing broadcast TV technology and standards and their evolution. The reasons are several. First, most of the same technologies and standards apply to interactive videoconferencing and multimedia. Second, although broadcast DTV and CATV do not necessarily have to follow the same rules of the road, it is likely that the CATV providers will carry broadcast TV signals in a digital format since they are upgrading their networks to digital anyway. The FCC has imposed must-carry rules on both the CATV and the DBS (Direct Broadcast Satellite) providers, thereby forcing them to carry local broadcast TV channels and to fairly compensate the local stations for that content. Considering that service providers are deploying broadband digital local loops in support of the triple play of voice, data, and TV, that the backbones are digital, that data, image, and movie content are digital, and that the TV broadcast signal will be digital, it is pretty clear that the digital circle will be complete very soon.
12.5 BANDWIDTH AND COMPRESSION
The bandwidth required for video transmission is very significant and is affected by frame rate, resolution, color depth, aspect ratio, and audio. Analog broadcast-quality TV requires about 6 MHz, of which the signal occupies about 4.5 MHz. In the United States, broadcast channel 2 is assigned frequency band 54-60 MHz, channel 3 is assigned the 60-66 MHz band, and so on, to channel 69, which is assigned the band 800-806MHz. (Note: There is no channel 1, but that story is too long and arcane for this work.) As little can be done to compress an analog signal, analog TV technology is rapidly being replaced with digital technology. Analog airwave broadcast signals in the United States will be turned off in 2009 and CATV networks are being converted quickly. Analog TV is doomed.
Digital video, on the other hand, can be compressed fairly easily. Because uncompressed, broadcast-quality digital video requires between 90 and 270 Mbps, compression is critical. Without compression, a 1-Gbps fiber-optic network could accommodate no more than 11 digitized NTSC channels. To digitize and compress the video information stream, the analog video and data signal first must be digitized through the use of a codec. Clearly, it is possible to reduce the amount of bandwidth required to transmit digital video and the amount of memory required to store it by reducing the frame rate, resolution, or color depth. However, the result is less than pleasing. In order to maintain the quality of the video presentation, therefore, the data must be compressed by using an appropriate and powerful data compression algorithm.
Lossless compression enables faithful reproduction of the video signal, with no data loss, although compression rates are in the range of only 10: 1. Lossy compression tends to produce artifacts, which are unintended and unwanted visual aberrations in the video image that often show up as jagged blockings or tiling effect known as aliasing, banding of colors, white spots, and even dropped frames. Although these artifacts result in a degraded picture, this lossy approach achieves compression rates up to 200: 1 [14]. Actually, compressed video currently can be transmitted with quite acceptable quality at T1 speeds of 1.544 Mbps or less. MPEG, for example, uses lossy compression in the form of Discrete Cosine Transform (DCT). A number of steps are involved in video compression, including filtering, color-space conversion, scaling, transforms, quantization and compaction, and interframe compression.
· Filtering, also known as image decimation, reduces the total frequency of the analog signal through a process of averaging the values of neighboring pixels or lines. For example, adjoining black-and-white pixels become gray pixels. Taps are the number of lines or pixels considered in this process; MPEG, for example, uses a seven-tap filter.
· Color-space conversion, also known as color sampling, simply involves the reduction of color information in the image. As the human eye is not highly sensitive to slight color variations, the impact is not noticeable. Black and white, however, are prioritized because the human eye is very sensitive at that level.
· Scaling addresses the creation of the digital image according to the presentation resolution scale associated with the display device. Rather than digitizing the video signal in large scale, the codec is tuned to the scale of presentation in terms of horizontal and vertical pixels, thereby reducing the amount of data that must be digitized. In consideration of this factor, the aspect ratio must be standardized.
· Transforms convert the native two-dimensional video signal into data dimensions. Although they are beyond the scope of this book, the various approaches include Discrete Cosine Transform (DCT), vector quantization, fractal transform, and wavelet compression.
· Quantization and compaction encoding simply reduce the number of bits required to represent a color pixel. Compaction techniques include run-length encoding, Huffman coding, and arithmetic coding.
· Interframe compression considers and eliminates redundant information in successive video frames. The background of a movie scene, for example, might not change, even though the actors move around the set. While the motion of the actors must be reflected, the background need not be retransmitted over the network. Rather, it can be compressed out of each frame until background changes must be reflected. At the point of decompression at the receive end of the signal, the unchanged background can simply be reinserted over and over again.
12.6 VIDEO STANDARDS
According to an anonymous but very wise man, "The nice thing about standards is that there are so many from which to choose!" Video compression standards are no exception—they are numerous and incompatible. Early standards were developed for specific purposes, such as motion picture production (MPEG) and photographic editing (JPEG). The ITU-T became involved in the development of international standards toward Broadband-ISDN (B-ISDN). In the videoconferencing world, numerous proprietary standards have been developed that require that the network involve equipment from only a single vendor.
As of late, a number of formal standards have developed. Those video compression standards of significance include P × 64, JPEG, and MPEG and its variations. Virtually all manufacturers of significance have embraced these standards, incorporating one or more of them into their systems alongside a proprietary compression technique. Generally involving P × 64, this approach provides at least a minimum level of communication between systems of disparate origin. The world of desktop videoconferencing is much more parochial; proprietary solutions long predominated, yielding incompatibility all too often [15]. As always is the case in networking, the highest common denominator rules. Lacking a common denominator, anarchy rules. The common denominator currently is P × 64, a standards-based solution that virtually all manufacturers support.
12.6.1 P × 64
P × 64 is an ITU-T standard designed to support videoconferencing and various levels of bandwidth, in p increments of 64 kbps up to a maximum of 2.048 Mbps (E1). Note: In P × 64, p = 1-30 channels of 64 kbps, with the maximum being the 30 bearer channels supported by E1. P × 64 specifies various frame rates and resolution levels. Most manufacturers of videoconferencing equipment support P × 64 as a common denominator, although they each prefer their own proprietary standards in promotion of their own equipment and unique feature sets. P × 64 more correctly is known as H.320 and is sometimes referred to as H.261, which is the specific ITU-T videocoding standard. The video formats include Common Intermediate Format (CIF), which is optional, and Quarter-CIF (QCIF), which is mandatory in compliant codecs. H.261 CIF supports 352 × 288 = 101,376 pixels per frame and 30 fps, although lower frame rates also are supported. QCIF supports 176 × 144 = 25,344 pixels per frame, exactly one-fourth the resolution of CIF.
Actually, H.261 is an element of the ITU-T H.320 umbrella standard for video telephony over circuit-switched ISDN and addressing narrowband visual telecommunications systems and terminal equipment. Related ITU-T standards include H.221, which defines a frame structure in support of audiovisual teleservices in 64-kbps channels, and H.222, which defines the frame structure for such services in an ISDN environment. While these umbrella standards truly are international in nature, the U.S. and European camps unfortunately (and not surprisingly) are divided over certain implementation aspects of the ITU-T standards. Specifically, those differences deal with the manner in which audio and still-frame graphics are handled. For graphics, for example, the Europeans have adopted JPEG (ITU-T T.81), which is compatible with most PC-based graphics software. That approach is totally incompatible with the H.261 standard implemented in the United States, although it does provide a bridge to true multimedia applications [16]. H.320 is discussed in greater detail later in this chapter.
12.6.2 Joint Photographic Experts Group
The ISO and ITU-T jointly developed Joint Photographic Experts Group (JPEG) as a compression standard for editing still images as well as color facsimile, desktop publishing, graphic arts, and medical imaging. A symmetrical compression technique, JPEG is equally expensive, processor intensive, and time consuming in terms of both compression and decompression. Motion JPEG is used in the editing of digital video. JPEG is not appropriate for video transmission, as the compression ratio is in the range of only 20: 1-30: 1. Therefore, JPEG transmission in support of videoconferencing requires bandwidth in the range of 10-240 Mbps, which is far too bandwidth intensive.
12.6.3 Moving Pictures Experts Group
MPEG standards are several and still in final development stages. MPEG standards provide very high compression levels and excellent presentation quality. MPEG is a joint technical committee of the International Organization for Standardization (ISO) and the International Electrotechnical Commission (IEC). MPEG offers the critical advantage of asymmetric compression and decompression. While MPEG compression is time consuming and expensive, the decompression process is rapid and involves relatively inexpensive equipment. MPEG compression is as high as 200: 1 for low-motion video of VHS quality, and broadcast quality can be achieved at 6 Mbps. Audio is supported at rates from 32-384 kbps for up to two stereo channels.
· MPEG-1 was standardized in November 1992 as ISO/IEC IS (International Standard) 11172; it provides VHS quality at 1.544 Mbps and is compatible with single-speed CD-ROM technology. In fact, it was designed as the standard for storage and retrieval of moving pictures and audio on storage media such as Compact Disc (CD). MPEG-1 integrates synchronous and isochronous audio with video and permits the random access required by interactive multimedia applications such as video games. Intended for limited-bandwidth transmission, it provides acceptable quality and output compatible with standard televisions. Current applications include video games, video kiosks, video-on-demand, and training and education. MPEG-1 supports video compression of about 100: 1. MPEG-1 standards are the basis for the MP3 (MPEG-1 audio layer 3) audio data encoding system. Although MP3 can produce CD-quality audio at compression ratios up to 12: 1, most MP3 codecs sacrifice considerable quality through compression at ratios as high as 25: 1.
· MPEG-2, defined in November 1994 as ISO/IEC IS 13818, is the standard for DTV. MPEG-2 was conceived as a broadcast standard for interlaced images at 720 × 480 pixels at 30 fps and requiring bandwidth at 4-100 Mbps over transmission facilities (e.g., fiber optics, hybrid fiber/coax, and satellite) capable of such support. While MPEG-2 requires much more bandwidth than MPEG-1, it provides much better resolution and image quality and at much greater speed. MPEG-2 has found application in Direct Broadcast Satellite (DBS) services, also known as Direct Satellite Systems (DSS). Such services employ Ku-band satellites and Very Small Aperture Terminal (VSAT) dishes in competition with CATV, running MPEG at rates of about 3 and 7.5 Mbps [16]. In a convergence scenario, MPEG-2 is the standard of choice, supporting compression rates as high as 200: 1, depending on content specifics. MPEG-3, designed for HDTV application, was folded into MPEG-2 in 1992.
· MPEG-4 Version 1 was approved in October 1998 as ISO/IEC IS 14496 and as the standard for multimedia applications. Version 2 was approved in 1999. MPEG-4 is a low-bit-rate version intended for application in broadcast TV, videophones, and mobile phones and other small hand-held devices. In a wireless context, MPEG-4 is designed for IMT 2000 applications that transmit at a rate of up to 384 kbps upstream and receive at a rate of up to 2 Mbps downstream, although rates of 64 kbps and lower are supported. Encompassing both client/server and mass storage-based playback scenarios, MPEG-4 deals with the coded representation of audiovisual objects, both natural and synthetic (e.g., computer generated), and their multiplexing and demultiplexing for transmission, playback, and storage.
· MPEG-7, approved in September 2001, officially is known as the Multimedia Content Description Interface. MPEG-7 is intended to be the content representation standard for multimedia information search, filtering, management, processing, and retrieval. MPEG-7 essentially is metadata standard based on XML (eXtensible Markup Language) for describing multimedia content features in order that one can easily search for multimedia content on the Web. (Note: According to the ISO, there apparently is no known reason for the failure to expand the logical numbering sequence and, instead, to jump to MPEG-7 and then to MPEG-21.)
· MPEG-21, officially known as Multimedia Framework, is an ongoing effort to determine how various multimedia components fit together and to identify new multimedia infrastructure standards that may be required. MPEC-21 also deals with issues of content identification, description, and security. In large part, the focus of MPEG-21 is on the protection of intellectual property through security mechanisms designed to prevent unauthorized access and modification of mul-timedia content.
12.7 INTERNET PROTOCOL TELEVISION (IPTV)
Several forces have concerned to create a great deal of interest in TV distribution over IP networks. There is an undeniable and irreversible movement towards IP, in general. Telephone companies are aggressively deploying broadband local loop technologies in the local loop to support triple-play services comprising voice, Internet access, and TV. While copper-based ADSL has limited ability to support TV, VDSL certainly can support it over short distances. The telco trend is more towards optical fiber in the loop, either directly from the Central Office (CO) to the premises in a Fiber-To-The-Premises (FTTP) Passive Optical Network (PON) scenario or a hybrid Fiber-To-The-Neighborhood (FTTN) scenario that terminates fiber in a neighborhood node and uses ADSL or VDSL over a short length of embedded copper cable from the node to the premises. Even full-rate ADSL supports enough bandwidth for at least one compressed video channel. VDSL and PON both support multiple channels.
The telco IPTV approach differs considerably from that of traditional CATV networks. CATV networks support analog TV over 6-MHz channels allocated from the total 850-MHz spectrum typically supported over coax or hybrid fiber/coax cable plant. The number of channels clearly is limited to the number of 6-MHz RF channels that can be derived from the available spectrum. As CATV networks are configured as point-to-multipoint networks, each channel is delivered to every connected subscriber premises. The IPTV approach delivers each channel only to the subscriber premises that select to view it, whether one or hundreds do so. Over a PON local loop, the TV channel occupies a separate wavelength from the head end to the premises. In an FTTN scenario, the IPTV signal is converted from optical to electrical format at the neighborhood node and travels over Unshielded Twisted-Pair (UTP) copper cable plant the last few hundred meters or so. AT&T's Project Lightspeed, for example, calls for 20–25 Mbps to each of 300–500 subscriber premises connected to its fiber-optic loops. Verizon's FiOS PON network offers 5, 15, and 30 Mbps, depending on subscriber preferences. Either network offers bandwidth sufficient to support multiple DTV and even HDTV channels, properly compressed. At the moment (9:05 A.M. PDT, May 21, 2007), however, the telco networks run video in uncompressed analog format over a separate wavelength at 1550 nm. The eventual conversion to IPTV speaks to compressed DTV, in packet format, likely still but not necessarily over a separate wavelength.
12.8 THE H.320 FAMILY OF MULTIMEDIA STANDARDS
The ITU-T has developed a number of Standards Recommendations for videote-lephony and multimedia communications. These recommendations fall under the umbrella of H.320, which was defined in 1990 for systems operating over Narrowband ISDN (N-ISDN). Table 12.3 details the range of standards under the H.320 umbrella. Each standard was developed for a specific network environment and includes standards for video and audio coding, signaling and control, and multipoint control units.
|
Table 12.3: Overview of ITU-T Videotelephony and Multimedia Standards Open table as spreadsheet |
|||||
|
Standard |
H.320 |
H.321 |
H.322 |
H.323, V1/V2 |
H.324 |
|
Approval date |
1990 |
1995 |
1995 |
1996/1998 |
1996 |
|
Network |
N-ISDN, PSTN |
B-ISDN, PSTN, ATM LAN |
Packet network with guaranteed bandwidth (e.g., IsoEthernet) |
Packet Network, with no guaranteed bandwidth (e.g., Ethernet, Token Ring, and Internet) |
Analog PSTN |
|
Video compression |
H.261, H.263 |
H.261, H.263 |
H.261, H.263 |
H.261, H.263 |
H.261, H.263 |
|
Audio compression |
G.711, G.722, G.728 |
G.711, G.722, G.728 |
G.711, G.722, G.728 |
G.711, G.722, G.728, G.723, G.729 |
G.723 |
|
Multiplexing |
H.221 |
H.221 |
H.221 |
H.225.0 |
H.223 |
|
Control |
H.230, H.242 |
H.242 |
H.230, H.242 |
H.245 |
H.245 |
|
Multipoint |
H.231, H.243 |
H.231, H.243 |
H.231, H.243 |
H.323 |
|
|
Data |
T.120 |
T.120 |
T.120 |
T.120 |
T.120 |
|
Communications interface |
I.400 |
AAL I.363, ATM I.361, PHY I.400 |
I.400 and TCP/IP |
TCP/IP |
V.34 modem |
|
Source: [17,18]. |
12.8.1 H.320
Also known as P × 64, H.320 supports videoconferencing and multimedia communications over N-ISDN B (Bearer) channels at bit rates from 64 kbps to 1.920 Mbps in increments of 64 kbps. Video compression makes use of H.261, which supports image resolution at several levels. As discussed earlier in this chapter, the optional CIF supports resolution of 352 × 288 pixels, and the mandatory QCIF supports resolution of 176 × 144 pixels. Frame rates are 30 fps or lower. Audio coding and compression recommendations include G.711, which is PCM at 64 kbps, thereby requiring a full ISDN B channel. G.722 is high-fidelity ADPCM transmitting up to a 7-kHz range and operating at 48/56/64 kbps. G.728 specifies LD-CELP running in the 3-kHz range and compressing voice at 16 kbps. The balance of the specification (H.221, H.230, and H.242) addresses techniques for call setup and teardown, data framing and multiplexing, and various other operational and administrative functions [17]. Table 12.4 details these various recommendations; the audio compression element largely is a restatement of the data provided in Table 10.3, which supports the discussion of packet Voice over Frame Relay (VoFR).
|
Table 12.4: H.320-Related Standards Recommendations Open table as spreadsheet |
|
|
ITU-T Standard Recommendation |
Description |
|
G.711 |
Pulse Code Modulation (PCM) voice coding at 64 kbps |
|
G.722 |
Adaptive Differential Pulse Code Modulation (ADPCM) voice coding and compression of high-fidelity 7-kHz voice at64/56/48 kbps |
|
G.723 |
Dual-rate speech coder at 5.3 and 6.3 kbps for multimedia communications |
|
G.728 |
Low-Delay Code-Excited Linear Prediction (LD-CELP) coding and compression of 3.3-kHz voice at 16 kbps |
|
G.729 |
Conjugate-Structure algebraic Code-Excited Linear Prediction(CS-CELP) voice coding and compression at 8 kbps |
|
H.221 |
Frame structure for channel of 64–1920 kbps in audiovisual teleservices |
|
H.223 |
Multiplexing protocol for low-bit-rate multimedia communication; annexes address mobile communications over low, moderate, andhighly error prone channels |
|
H.225 |
Call signaling protocols and media stream packetization for packet-based multimedia systems |
|
H.230 |
Frame synchronous control and indication signals for audio visual systems |
|
H.242 |
System for establishing communications between audiovisual terminals using digital channels up to 2 Mbps; addresses call setupand teardown, in-band signaling and control, and channelmanagement |
|
H.245 |
Call control procedures for multimedia communications |
|
H.261 |
Video codec for audiovisual services at p × 64 kbps |
|
H.263 |
Video coding for low-bit-rate communication at rates less than64 kbps |
|
T.120 |
Multipoint transport of multimedia data |
|
Source: ITU-T. |
12.8.2 H.321
H.321 is the ITU-T Standard Recommendation for the adaptation of H.320 visual telephone terminals to B-ISDN environments. B-ISDN depends on ATM switching, which offers the considerable advantage of guaranteed Quality of Service (QoS), as discussed in Chapter 10. H.321 also involves H.310, which is the recommendation for broadband audiovisual communications systems and terminals.
12.8.3 H.322
H.322 is the ITU-T Recommendation for visual telephone systems and terminal equipment for LANs that provide a guaranteed QoS. This specification is limited to IsoEthernet (Chapter 8), which fact renders it little more than a historical footnote.
12.8.4 H.323
H.323 is the ITU-T Recommendation for packet-based multimedia communications systems. Annex D describes real-time facsimile. The recommendation addresses LANs that do not provide a mechanism for guaranteed QoS. H.323 also is used for service over the Internet and other IP-based networks, as illustrated in Figure 12.1. H.323 offers the advantage of supporting various compression techniques for packet-based voice communications, which can yield much more efficient utilization of network resources than does the traditional G.711 approach of PCM over circuit-switched networks. Interoperability of products (e.g., terminals and switches) can be achieved across the LAN and WAN domains. As H.323 is not linked to any specific hardware device or operating system, it can be deployed in a wide variety of devices, including PCs, telephone sets, and cable modems and set-top boxes. H.323 supports multicast communications, thereby avoiding the requirement for specialized Multipoint Control Units (MCUs) in a network where routers assume the responsibility for replicating packets. Version 2, ratified by the ITU in September 1998, provides a means for encryption, includes mechanisms for call transfer and call forward, supports URL-style addresses, and provides the ability for endpoints to set QoS levels through Resource Reservation Protocol (RSVP). The four major components specified for H.323 include terminals, gateways, gatekeepers, and Multipoint Control Units (MCUs).
· Terminals are the client endpoint devices on the LAN. While all terminals must support voice, data and video are optional. H.245 must be supported for negotiation of channel usage and capability. Q.931 is required for signaling and control. The Registration/Admission/Status (RAS) protocol communicates with the gatekeeper. Sequencing of audio and video packets is supported through Real-time Transport Protocol/Real-time Transport Control Protocol (RTP/ RTCP). Endpoints can set QoS levels through Resource reSerVation Protocol (RSVP), although RSVP is unusual. Terminals optionally may include video codecs, T.120 data conferencing capabilities, and MCU functionality.
· Gateways are optional elements in the H.323 environment used for various levels of protocol conversion. A gateway comprises a Media Gateway (MG) and a Media Gateway Controller (MGC), which commonly are physically distinct devices. The MG serves as a protocol converter between devices and networks that have native H.323 capability and those that do not. The gateway also may translate between audio, video, and data formats. Finally, the gateway may perform signaling conversions between the H.225 packet protocol and external protocols such as Signaling System 7 (SS7) and Q.931. Alternatively, signaling conversions may be performed by gatekeepers, call processors, or session border controllers.
· Gatekeepers act as the central points in H.323 zones (i.e., zones of control). Endpoints may communicate directly, in either a unicast or a multicast environment, if no gatekeeper is present. If a gatekeeper is present, all endpoints in its zone must register with it. The gatekeeper performs the function of admission control, determining if devices are authorized to connect and if there is sufficient bandwidth to support the call. Gatekeepers serve to translate LAN addresses into IP or Internetwork Packet eXchange (IPX) addresses, as defined in the RAS specification. Gatekeepers also can act to route H.323 calls through gateways, if necessary, and monitor the network bit rate capacity, with the ability to deny access to a session if programmable bandwidth thresholds have been reached or exceeded. Gatekeepers also can perform certain administrative functions, such as accounting, billing, directory, and collecting network usage data. Gatekeepers may be distinct network elements, or gatekeeper functionality can be incorporated into (MCUs).
· Multipoint Control Units (MCUs) support conferencing among three or more participating terminals. The MCU comprises a Multipoint Controller (MC) and optional Multipoint Processors (MPs). The MC is responsible for call control negotiation to achieve common levels of communication. The MP may process either a single media stream or multiple media streams, depending on the nature of the conference.
Figure 12.1: H.323 networking over IP-based network
12.8.5 H.324
H.324 is the ITU-T Recommendation for low-bit-rate multimedia communication over the analog PSTN through V.34 modems. As such, modems are limited to maximum transmission rates of 28.8 kbps. Voice must be highly compressed in order to make room for video and other visual information streams [17–20].
12.8.6 T.120
The ITU-T Recommendation for the multipoint transport of multimedia data is T.120, which data can include whiteboarding or binary files. This series of recommendations supports a broad range of underlying network technologies and can work either alone or under the H.320 umbrella. T.120 is entirely platform independent and can run in a variety of network environments, involving either reliable or unreliable data transport. Unicast and multicast modes both are supported [17,21].
12.9 SESSION INITIATION PROTOCOL
The Internet Engineering Task Force (IETF) defined Session Initiation Protocol (SIP) in its RFC 2543 (March 1999) as an application layer (Layer 7 of the OSI Reference Model) signaling protocol for establishing, modifying, and terminating multimedia sessions or calls over an IP network. Finally approved in June 2002 in RFC 3261, which obsoleted RFC 2543, SIP is a modular component of IP telephony, although it can function over any network. SIP offers considerable advantages over H.323, although it can be used in conjunction with H.323 where appropriate. H.323 commonly is criticized as being too slow in establishing sessions, that is, setting up packet calls. An H.323-compliant client initiating a call must query a gatekeeper for the address of a new destination device. Once the gatekeeper has provided that address, the originating client establishes the session by using the H.225 signaling protocol, and the two clients negotiate features and call control procedures by using H.245. All of this takes time. In fact, it can take much more time than is required to set up a call through the conventional circuit-switched PSTN. The exact amount of time is sensitive to network specifics, and you must consider the fact that the Internet is a network of many networks, each of which is of uncertain nature in many respects. H.323 also is criticized for being overly complex and highly centralized.
SIP addresses all of these problems, and it was built specifically for an IP environment in which intelligence is highly decentralized in a large number of client agent servers. SIP identifies clients through a hierarchical URL similar to an e-mail address, e.g., SIP:[email protected]. (Don't try it, as I'm not SIP-compliant at the moment.) There are two ways that the calling client can initiate the call. If the SIP address of the destination SIP client is known, the calling client simply sends the destination client an invite message, in care of (1) a local proxy server. The proxy server sends the invite message to the distant proxy server, inviting the destination endpoint to join the session and providing it with enough information to do so. If the SIP address of the destination SIP client is unknown, the calling proxy server sends (2) the invite message to a redirect server, which consults (3) the location server for address information. The redirect server passes (4) that information to the calling proxy server, which then issues (5) an invite message to the distant proxy server, including the information required to join the call. If the call is to a call center, such information might include a request to employ H.261 video, G.728 audio, and Japanese as the preferred language. The proxy server on the receive end might consult (6) an optional SIP location server on the receive end to determine the exact location of the called client and connect (7) the call (i.e., ring the multimedia PBX). This approach, as illustrated in Figure 12.2, is a lot simpler and faster than the back-and-forth process involved in H.323, although layers of complexity are being added as the standards process works to enhance SIP to match H.323 and PSTN functionality.
Figure 12.2: IP networking using SIP trunks and SIP over PSTN trunks
Once the called client receives the invitation to join the session, it can either accept the call or forward it to a messaging system or a user, perhaps a Japanese-speaking call center agent. Assuming that the call is a multimedia call comprising both video and voice, the called client (or messaging system) can elect either to accept the composite call or to accept only one of the data streams, perhaps rejecting the video call but accepting the voice call. SIP also supports call forking, or splitting, so that several client extensions can be rung at once. H.323 does not have the same call-forwarding capabilities, although there is a follow-me feature that is similar. H.323 does not support constituent call separation or call forking. The big drawback to SIP is its lack of availability. H.323 is the incumbent protocol, and you know how hard it is to unseat an incumbent. As long ago as early 2001, however, several major carriers announced the commercial availability of limited SIP-based Voice over IP (VoIP) services and their plans to extend that capability in the future. Also particularly noteworthy was Microsoft's decision to incorporate SIP in Windows XP, Windows CE 4.0, Windows.NET server, and other devices embedded with XP [22–29].
A number of carriers now offer SIP trunks that connect the end-user premises directly to the SIP-enabled IP network, as illustrated in Figure 12.2. This approach certainly is the purest, as the enterprise-level SIP clients and servers interconnect directly through the SIP-enabled IP network. The alternative is for the enterprise user to gain access to the IP network through the TDM-based PSTN, with multiple Media Gateways (MGs) making the IP/TDM/IP conversions.
12.10 H.248: MEDIA GATEWAY CONTROL
The Media Gateway Control (Megaco) protocol is a joint standardization effort of the ITU-T (H.248) and the IETF (RFC 3525). Megaco, as it is known at the IETF, evolved from Simple Gateway Control Protocol (SGCP) and Media Gateway Control Protocol (MGCP). Megaco defines the call control protocols employed in a physically decomposed gateway with subcomponents distributed across multiple devices that may be in multiple physically distinct locations. Those subcomponents take the form of a Media Gateway (MG) and a Media Gateway Controller (MGC), also known as a softswitch or a call agent. A single MGC can control a large number of MGs, each of which is optimized for a particular gateway application function to convert the media format between a packet network and another form of network. Examples include interfaces that accomplish media conversion between IP packet format and an analog PSTN, or DS-0 format in a T/E-carrier-based PSTN interface, or an ATM network, or perhaps a device such as a PSTN-based voice processor.
The call control and signaling logic are centralized in the MGC and can include features such as dial tone, collect dialed digits, call hold, call transfer, call forward, and call conference. Feature changes are made only to the MGC, which simplifies the process of administration. The MGC signals the MGs, which then execute the feature commands and process the call, performing gateway functions as required to interface the incompatible networks or network elements. There is a master/ slave relationship between the centralized MGC and decentralized MGs, much like that of a traditional telco network, except for the fact that the MGs that execute the features and perform the switching are distributed across the network. This is in sharp contrast to SIP, which is even more highly decentralized [28–30].
12.11 VIDEOCONFERENCING SYSTEMS
During 1930 two-way Television was demonstrated over a circuit connecting the Bell Telephone Laboratories with the headquarters of the American Telephone and Telegraph Company. Persons in booths two miles apart were enabled to see moving images of each other while they conversed. Television, however, is still in the development stage, and is not available on a commercial basis.
Things Worth Knowing about The Telephone, American Telephone and Telegraph Company, 1931
Videoconferencing systems consist of cameras, monitors, video boards, microphones, speakers, and software. Videoconferencing can be accomplished in the workplace over a LAN. While it generally is more effective, easier, and less costly to walk across the hall and hold a face-to-face meeting, contemporary LANs often extend across multiple floors and even multiple buildings in a campus setting. Videoconferencing over the WAN clearly offers great benefits in terms of reduced cost and increased availability for meetings. Systems for videoconferencing can be quite substantial or can be PC based. Videoconferencing has increased significantly over the past few years as the cost of equipment and bandwidth have decreased—systems are made up of room systems, rollabout systems, and PC-based systems.
· Room systems are complex and quite expensive systems intended for videoconferencing among groups of people. While a specially designed room and equipment easily could cost $ 250,000 in the 1980s, a room system can be configured for less than $ 25,000 today. AT&T, MCI, and Sprint have provided room-based conferencing services for years from select locations, and they also provide network-based videoconferencing services for large corporations. Such service offerings include access and transport services as well as network-based MCUs. While many U.S. Incumbent Local Exchange Carriers (ILECs) in the late 1980s or early 1990s announced plans for video dial tone, those announcements clearly were way ahead of both the technologies and the market, and all of those plans were abandoned. The proliferation of fiber optics in the network core and advent of PON in the local loop now make such a service technically feasible, at least in terms of transmission capacity. Kinko's Copy Centers (now FedEx Kinko's) began deploying videoconferencing services in 1994 and currently offers such services at 150 locations in the United States and Canada. Kinko's has improved quality to 30 fps at 384 kbps through an alliance with Sprint for high-speed access and transport.
· Rollabout systems essentially are portable and much less expensive versions of room systems and also are intended largely for group-to-group conferencing. Such systems account for the preponderance of the stand-alone equipment market.
· PC-based systems, also known as desktop or appliance-based systems, are intended for person-to-person conferencing. PC-based systems are enjoying increased popularity, with the systems generally working over LANs or ISDN circuits. Analog transmission over the WAN also is accommodated by some systems, through V.34 and V.34+ modems at speeds up to 33.6 kbps. As you might expect, the cost of the system and the network are directly related to the frame rate supported, the image resolution, and the sophistication of the compression technique employed. Inexpensive PC-based systems can be had for as little as several hundred dollars today. The cost is that of camera and codec, the latter of which fits into the expansion slot of a PC or Macintosh computer, with the monitor serving as the video presentation device. Inexpensive PC-based systems are used for Internet videoconferencing, although quality is poor due to latency and data loss associated with packet switching over the public Internet.
12.12 VIDEOCONFERENCING EQUIPMENT
Videophones originated with the AT&T Picturephone, which was demonstrated at the New York World's Fair in 1964. Never intended for practical application, the Picturephone was extremely bandwidth intensive, requiring bandwidth of about 90 MHz and weighing about 26 lb [6]. During the 1980s, AT&T, BT, and others developed videophones that sold for less than $ 1000. As the cost was high, as each party was required to have a videophone of the same manufacture, and as the picture quality was poor (2 fps), videophones were stunning failures.
Regardless of the nature of the system, some combination of hardware and software is required. Videoconferencing equipment includes transmit (camera) and receive (display) equipment that operate in concert with and through various intermediate devices to format the signal properly and otherwise treat it for effective transmission over a network. Those intermediate devices include codecs, inverse multiplexers, servers, and control units. Figure 12.3 presents a simple videoconferencing arrangement.
· Codecs (coders/dec oders) accomplish the process of digitizing, or coding, the analog signal on the transmit side and decoding it on the receive end. The codecs also accomplish the process of data compression and decompression, according to the specifics of the compression algorithm used. Additionally, codecs may include encryption features for security purposes and a mechanism for synchronizing the audio and video elements of the transmission.
· Inverse Multiplexers (inverse MUXs) are used in commercial videoconferencing systems where dedicated bandwidth is not available for relatively bandwidth intensive communications. An inverse MUX splits the video signal into two or more component parts that are transmitted over separate circuits or, perhaps, separate channels of multiple multichannel circuits (e.g., T1). The inverse MUX on the receiving end reassembles and resynchronizes the complete video signal for proper presentation.
· Servers are extremely high capacity storage devices, containing many gigabytes or even terabytes of memory. Servers store video and audio data for delivery to clients on demand.
· Multipoint control units are digital switching and bridging devices that support multipoint videoconferencing, with up to 28 parties (sites) commonly supported. MCUs must be compatible with the compression standards employed with the codecs. H.231, for example, describes ITU-T MCU standards, and T.120 describes generic data conference control functions. MCUs may be found in the carrier network in support of a carrier videoconferencing service or on the end-user premises in support of a videoconferencing network based on leased lines.
Figure 12.3: Simple videoconferencing network employing cameras, codecs, and monitors
12.13 WAN VIDEOCONFERENCING NETWORKS
Video networking can be accomplished over a number of facilities and service offerings, depending on the application and the amount of bandwidth required. Analog circuits will support videoconferencing at low speeds, although the results are less than completely pleasing. The failed videophones offered by AT&T, BT, and others made use of dial-up analog circuits or ISDN circuits. As always is the case, digital circuits offer better performance than do analog circuits.
ISDN circuits are preferable to analog circuits because they provide more bandwidth and better transmission quality. However, the higher cost and lower availability of ISDN have slowed the acceptance of videophones based on ISDN technology in the United States. Note that in Europe and Japan ISDN is not particularly expensive and is much more widely available. Switched 56/64-kbps circuits can be used for videoconferencing—generally aggregated or bonded to provide multiple channels. Switched 384-kbps connectivity can be provided on the basis of fractional DS-1 through ISDN PRI channels in a channel group known as HO or over ADSL.
DS-1 facilities support full-motion, high-quality videoconferencing over dedicated networks at rates up to 2.048 Mbps for E-1 and 1.544 Mbps for T1. However, such facilities are costly and not widely dedicated to such applications. Large user organizations with dedicated leased-line T/E-carrier backbone networks make highly effective use of videoconferencing; the video communications contend with voice and data for network access through intelligent MUXs.
Broadband networks are much more capable of supporting the demands of videoconferencing. Frame Relay supports video, although that clearly is not the primary reason for its existence. While Frame Relay performs well under normal circumstances, it is likely to yield herky-jerky video should a poorly designed network suffer severe congestion. ATM was positioned for some years as the network technology of choice in a convergence scenario, at least in the backbone. ATM offers tremendous bandwidth over fiber-optic or hybrid fiber/coax networks and in support of voice, image, facsimile, and data traffic as well as video. ATM offers the unique advantage of supporting all of these traffic types, simultaneously providing each application type with precisely the QoS it expects. Ultimately, ATM fell from grace due to its complexity and high cost. IP has now replaces ATM as the favored protocol du jour, not only for data and voice but also increasingly for video.
12.14 VIDEO OVER IP
I spent a good deal of ink in several previous chapters examining VoIP. Now, I want to shift the emphasis to the other VoIP, Video over I P, which is quite different from IPTV. Video, like voice, is stream oriented and isochronous in nature. That is to say that video, like voice, depends on a continuous flow of interrelated data from the transmitter across the network to the receiver. For that matter, video, in general, usually is a combination of video and voice, and I cannot think of an example of videoconferencing that is not characterized by the same combination. The video and voice information streams are layered together and carefully synchronized so that the voice audio matches the movement of the speaker's lips. Just like stand-alone voice, interactive videoconferencing is very demanding in terms of QoS. It is critical that latency is in the range of 150 ms or less, that jitter be in the range of 50 ms or less, and that packet loss be in the range of 1 percent or less. Issues of latency and jitter are not particularly critical in a one-way streaming video application, as the receiver has time to buffer the incoming data stream and adjust. In a real-time, two-way videoconference, however, these issues are very significant. That's quite a trick over an IP network, but it can be done—and increasingly is being done with quite satisfactory results.
Video is extremely bandwidth intensive. Although video can run at rates as low as 64 kbps, high-quality video requires a high frame rate, good color depth, and good resolution, and those dimensions require a lot of bandwidth. So, the more bandwidth available from end to end, the better the potential quality of the presentation, which of course is also sensitive to the quality of the equipment involved. Given the fact that bandwidth is always limited and considering that the weakest link in the network (i.e., the local loop) always rules, compression is a good idea for both the video and voice elements. H.263, for example, is the choice of many manufacturers for video compression and G.723 (ADPCM) for voice. Because video and voice are stream oriented and, therefore, are intolerant of latency and jitter, it is a good idea to prioritize the data stream throughout the packet network, and that is where Differentiated Services (DiffServ) comes in. Considering the fact that there always will be some level of jitter and loss across a highly shared packet network, it is a good idea to both prioritize packets and provide a mechanism for reconstructing and resynchronizing packets at the receiving end, and that is where Real-time Transport Protocol (RTP) comes into play. DiffServ and RTP are particularly important in both video over IP and voice over IP, as UDP (User Datagram Protocol), rather than TCP, is used at layer 4. UDP is less bandwidth intensive than TCP but provides no error correction mechanism. Rather, the application assumes all responsibility for error correction. There's no time for retransmission of errored or lost packets in a real-time stream-oriented application, anyway, so it is up to the application to deal with those issues as it sees fit, if at all. Video over IP can work and does work, but all of the right pieces have to be in place for it to work well.
That said, it may surprise you—and it certainly surprised me—to hear that one of the reasons behind the increase in IP videoconferencing is that it actually works better than ISDN. According to Chris DiFiglia, president of the Polycom User Group, the typical 384-kbps ISDN videoconference depends on getting six bonded ISDN B channels to work simultaneously, which can be difficult. DiFiglia states that "IP is a much more fault-tolerant network. With ISDN, if there's a dropped frame or picture the entire call is dropped. An IP world is more forgiving. The worst that would happen is that the video frame would freeze for a second" [31]. IP videoconferencing certainly is less expensive, as ISDN is notoriously costly, and the six B channels carry per-minute charges, which can be particularly significant for international calls. ISDN videoconferencing also requires fairly expensive software and terminal equipment or terminal adapters. I P, on the other hand, generally involves little additional cost.
|
|
|
12.15 MULTIMEDIA CONFERENCING
Software must be in place to support electronic text, image, audio, and video information in a multimedia conference. While the voice and video aspects of the conference are supported in a fairly straightforward manner, the real and distinct advantage of multimedia conferencing is that of enabling multiple parties to collaborate on textual and graphic documents. Special software enables each party to contribute to such documents, in collaboration with the other parties—hence the term collaborative computing. During such a collaborative session, the original text document is saved, while each party contributes changes that are identifiable as such, by contributor. Once the parties agree to the collaborative edits and enhancements, the entire text file is refreshed and saved.
Similarly, a design or a concept can be developed graphically and on a collaborative basis through whiteboarding, much as the parties would do on a physical whiteboard in a face-to-face meeting. Typically, each party to the conference has access to a special whiteboard pad and stylus which is used to draw. Each party can modify the initial drawing, with each individual's contribution identified by separate color. Again, and once the group has agreed on the final graphic rendition, the graphic is saved and all screens are refreshed.
The clear benefits of such a collaborative process, conducted on a logical basis over a WAN, include reduced travel time, reduced travel expense, and increased speed of collaborative effort. Even in a LAN environment, shoe leather is conserved and productive time is maximized. A number of online Web-based collaborative conferencing tools are now available, offering considerable benefits over audio-only conferencing.
12.15.1 Video and Multimedia Conferencing: Applications and Benefits
There can be little doubt about the value of enhancing a communication with visual information. Pictures add another and very important element to the process of learning and comprehension. Moving pictures enable us to see the person to whom we are talking, thereby creating a more natural and effective person-to-person communication. Taking it one step further, a truly collaborative effort is enhanced greatly when multiple persons can work on a document. Supporting all of this over a network can save a lot of money, time, shoe leather, and gasoline. In a February 2004 report, Yankee Group analysts cited a number of benefits for videoconferencing, particularly in a contemporary multinational, multicultural environment [32]:
· Fifty-five percent of the impact of communications comes from facial expressions and body language, and only 38 percent from vocal inflection. (Source: UCLA.)
· Attendees in face-to-face meetings retain 38 percent more information than those in audio-only meetings. (Source: Harvard University and Columbia University.)
· Face-to-face meetings increase the power or persuasion by 43 percent over audio-only meetings. (Source: 3M Company.)
· Attendees learn 200 percent more in face-to-face meetings than in audio-only meetings. (Source: University of Wisconsin.)
· Attendees absorb information as much as 40 percent faster in face-to-face meetings than with audio alone. (Source: Wharton School of Business.)
Add to these benefits of electronic face-to-face meetings the hard dollar savings in travel expense and employee travel time, and a compelling case can be made for electronic conferencing. The unfortunate events of September 11, 2001, added the element of personal safety to the list of forces driving renewed interest in videoconferencing. Serving to supplement, but never replace, face-to-face in-person collaboration, video and multimedia networking has a legitimate place in the networked world.
The development of videoconferencing has been much slower than many of us projected. Traditional system and network technologies never developed to the point that high-quality videoconferencing was either simple or affordable. The recent development of standards for IP video promises to overcome those issues of complexity and cost, given the high availability and reasonable cost of broadband capacity in both the core and the local loop. Large organizations just cannot deny its value, and the costs of the system technologies are becoming quite reasonable for video-intensive applications. As costs continue to drop, small-business, home business, and even consumer markets for multimedia equipment will expand greatly.
One thriving application is that of telemedicine, which supports consultation and even remote diagnosis and treatment. A number of projects have experimented successfully with this concept, largely in support of remote clinics. Through a videoconferencing system and network, a nurse or medical technician in a remote clinic can gain the assistance of a doctor, and even a specialist, located at a major urban hospital. The doctor can guide the technician through the process of diagnosis and treatment, viewing the patient over a videoconferencing network and perhaps viewing x-rays transmitted over the same high-quality, high-speed digital network. For that matter, a multipoint conference can be established so the physician might consult with distant colleagues on a particularly difficult diagnosis and treatment plan. Taking the scenario one step further, the physician can even guide the technician through an emergency surgical procedure. Prescriptions, clearly, can be transmitted electronically to a local pharmacy. While such an application scenario currently is a bit unusual, it is possible, has been accomplished, and is in daily use. Although the typical commercial enterprise might not find videoconferencing or multimedia networking to be a lifesaving application, they will be critical elements of the technology mix for those firms seeking to gain or maintain competitive advantage.
Another interesting application for videoconferencing is in the justice system. Judges around the country, including in my little town of Mt. Vernon, Washington, make use of videoconferencing systems for video arraignments and other court appearances required for suspected criminal defendants. The clear advantage is that a video communication between the judge and lawyers in the courtroom and the defendant in the jailhouse eliminates the costly and sometimes dangerous process of transporting the accused and convicted. Tom Swift would have been proud, indeed, of the evolution and application of his photo phone.
Yet another interesting potential application is that of voice-and video-enabled websites. Clearly, the customer's contact experience can be improved tremendously if he or she can click a videoconference button and connect to an agent to establish a videoconference. That assumes, of course, that both parties are properly prepared for the camera.
Despite increased bandwidth, improved compression, and all the various protocols and mechanisms that have improved, and will continue to improve, video quality over the years, the videoconferencing experiencing still leaves us flat—two dimensional, that is. Wouldn't if be nice if video were in a 3-D (three-dimensional) format? Well, there has been a lot of development effort expended to do just that, and there is at least one commercially available system. Teleportec has developed a system that creates a 3-D effect through the use of multiple cameras on the transmit side and on the receive side by projecting the image on a specially treated sheet of glass that acts as a beam splitter, refracting light at different rates. The system builds on the H.320 and H.323 standards and currently runs over ISDN or dedicated circuits. A full T1 is preferred for maximum quality, but 768 and 384 kbps provide acceptable quality [33].
|
|
|