Search

IPTV can be a new service performed on the Internet in that network transmission and streaming media technologies are getting mature. In this paper, IPTV system infrastructure of UTStarcom, key technologies deployed, and applications will be discussed and evaluated. The key technologies to achieve IPTV services include 1) codec and compression; 2) streaming media; and 3) broadband networks and access to such networks. The implementation of Media Switch IPTV system in Harbin CNC city network is also discussed.

For fiber-optic links (FOL), the sufficient time of repairing (removing & reinstalling) fiber-optic cables (FOC) and the natural ageing of optical fibers (OF) have been stipulating and propelling the problems of safe communication. The latters used to and are nowadays resolved by mean of either reserving line, cables, fibers, digital streams or improving quality of FOC grooming. In the latter case, the implementation of a prognostic control of FOC plays sufficient role in fault prediction on maintenance stage of FOL. The prognostic control is performed by mean of Remote Test Fiber Systems (RFTS) on the basis of both direct controlling OF parameters and indirect methods of FOC monitoring, including, being applied at Volgograd City Telephone Network, RLTP Method (Rate Loss Time Prediction) designed for copper/ steel media. Thus for example a prediction method on the basis of protective sheath monitoring results of FOC. This method is very effective when applied for buried long-haul FOC and in widespread use in Russia. One of the problems ofthe method performance is how to maintenance satisfactory sheath condition ofFOC. In general, the main trouble is how to localize faults on the protective sheath. There are however some classical techniques which are similar to the applied techniques in fault localization in copper media, but when applied for FOL, the techniques have specifics. The purpose of the paper is to analyze specifics ofthe classical technique implementation for fault localization ofprotective sheath of FOC.

This system adopted two-level structure that was composed of Cable Monitoring And Controlling Device (CMCD) and Cable Monitoring Center (CMC). CMC receives the alarm data via the MODEM and the telephone net. In this way, the functions of typing the fault point's map, alarming and the cable management can be picked up. CMCD takes the processor chip Atemega128 as the center of the system that adopted the alternating current and direct current on-line switching electricity-supply mode. Also, the system includes four groups of independent power, the relay and the optical isolation to separate the system from the monitoring cable and the telephone net. CMCD accomplishes the cable real-time monitoring, malfunctions auto-locating, telephone voice alarming, long-distance parameters modification, data up-loading, error verifying and the telecommunication room's environment monitoring and so on. The longest distance of monitoring cable is 20 km, and the precision is 1%.

The recent ITU-T Recommendation P.862, known as the Perceptual Evaluation of Speech Quality (PESQ) is an objective end-to-end speech quality assessment method for telephone networks and speech codecs through the measurement of received audio quality. To ensure that certain network distortions will not affect the estimated subjective measurement determined by PESQ, the algorithm takes into account packet loss, short-term and long-term time warping resulted from delay variation. However, PESQ does not work well for time-scale audio modification or temporal clipping. We investigated the factors that impact the perceived quality when time-scale modification is involved. An objective measurement of time-scale modification is proposed in this research, where the cross-correlation values obtained from time-scale modification synchronization are used to evaluate the quality of a time-scaled audio sequence. This proposed objective measure has been verified by a subjective test.© (2003) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

Signaling System 7 (SS7) is vital to signaling and control in America's public telephone networks. This paper describes a class of attacks on SS7 networks involving the insertion of malicious signaling messages via compromised SS7 network components. Three attacks are discussed in detail: IAM flood attacks, redirection attacks and point code spoofing attacks. Depending on their scale of execution, these attacks can produce effects ranging from network congestion to service disruption. Methods for detecting these denial-of-service attacks and mitigating their effects are also presented.

Converged network security is extremely challenging for several reasons; expanded system and technology perimeters, unexpected feature interaction, and complex interfaces all conspire to provide hackers with greater opportunities for compromising large networks. Preventive security services and architectures are essential, but in and of themselves do not eliminate all threat of compromise. Attack management systems mitigate this residual risk by facilitating incident detection, analysis and response. There are a wealth of attack detection and response tools for IP networks, but a dearth of such tools for wireless and public telephone networks. Moreover, methodologies and formalisms have yet to be identified that can yield a common model for vulnerabilities and attacks in converged networks. A comprehensive attack management system must coordinate detection tools for converged networks, derive fully-integrated attack and network models, perform vulnerability and multi-stage attack analysis, support large-scale attack visualization, and orchestrate strategic responses to cyber attacks that cross network boundaries. We present an architecture that embodies these principles for attack management. The attack management system described engages a suite of detection tools for various networking domains, feeding real-time attack data to a comprehensive modeling, analysis and visualization subsystem. The resulting early warning system not only provides network administrators with a heads-up cockpit display of their entire network, it also supports guided response and predictive capabilities for multi-stage attacks in converged networks.

This paper presents a remote state monitoring system designed for and installed on the Hongcaofang Crossroad Bridge in Chongqing, China. In this system, three kinds of sensor, one of which is new, are installed in the bridge to periodically collect strain and deflection information. To control the operation of the sensors, a local computer is integrated in the pier of the bridge. The local computer processes the data from sensors, records processed results, and sends the state information to a host computer through the local Public Service Telephone Network (PSTN) using an ordinary modem. At the other terminus, the host computer receives and analyzes the data, stores the history information, queries the health state, and extracts abnormity information regarding the bridge. With the interconnect technology available through the PSTN, real time state information can be obtained on command in the monitoring room far from the bridge. This on-line monitoring system operated on the Hongcaofang Bridge for over two years. This paper reviews some of the more important results regarding both the strain and the two-dimensional deflection of the bridge, and discusses the experience gained thus far.

For fiber-optic links (FOL), the sufficient time of repairing (removing & reinstalling) fiber-optic cables (FOC) and the natural aging of optical fibers (OF) have been stipulating and propelling the problems of safe communication. The latters used to and are nowadays resolved by mean of either reserving line, cables, fibers, digital streams or improving quality of FOC grooming. In the latter case, the implementation of a prognostic control of FOC plays sufficient role in fault prediction on maintenance stage of FOL. The prognostic control is performed by mean of Remote Test Fiber Systems (RFTS) on the basis of both direct controlling OF parameters and indirect methods of FOC monitoring, including, being applied at Vol-gograd City Telephone Network, RLTP Method (Rate Loss Time Prediction) designed for copper/steel media. Thus for example a prediction method on the basis of protective sheath monitoring results of FOC. This method is very effective when applied for buried long-haul FOC and in widespread use in Russia. One of the problems of the method performance is how to maintenance satisfactory sheath condition of FOC. In general, the main trouble is how to localize faults on the protective sheath. There are however some classical techniques which are similar to the applied techniques in fault localization in copper media, but when applied for FOL, the techniques have specifics. The purpose of the paper is to analyze specifics of the classical technique implementation for fault localization of protective sheath of FOC.© (2001) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

The International Maritime Satellite (Inmarsat) communication service can be used to provide a wide area network (WAN) connection for a local area network (LAN) to the home LAN from almost anywhere on the earth. The communication link of the WAN connection is achieved through the satellite network of Inmarsat and a land-earth digital telephone network, such as Integrated Services Digital Network (ISDN). We determined the throughput data rate of data transmission in the TCP/IP environment over the Inmarsat link and compared it with the throughput data rate over a null-modem link (assumed to be error free) for the WAN connection. We also used the protocol boosters that provide forward error correction for the TCP/IP environment to test the improvement of data transmission over the satellite link. Communication over satellite tends to be noisy and delay prone. However, our analysis shows that the Inmarsat satellite-ISDN link is so reliable that the data transmission rate is nearly an error-free link in the TCP/IP network, and the application of forward error correction on the network causes a reduction in throughput data rate.

Traditionally, voice has been transported using circuit switched networks, the Public Switched Telephone Network (PSTN) for example. However, driven by the ubiquity of the Internet and the development of low bit-rate digital voice codecs there has been increasing focus on using packet-switched networks for voice traffic. We focus on one such application. Our model comprises of a relatively slow packet link (between 1.5 and 5.0 Mbps) being utilized for voice traffic. Large number (of the order of 100) voice sources are multiplexed on this link. For such a link we obtain delay distributions seen by a voice source. Specifically, we exploit key characteristics of the model, such as the large number of sources to obtain a heavy traffic approximation for the system. The key result is that the delays can be well approximated by concatenation of two exponential distributions. We also provide valuable insight into how the delay distribution is connected with the statistical properties of the voice sources, in particular their correlation behavior. Our analytical results are validated with simulation results.

A real-time VLBI (Very Long Baseline Interferometer) system using conventional telephone lines is proposed. This system enables us to check if the VLBI network is alive by on-line manner, which is useful to reduce the loss of observation time caused by system malfunctions. In this system, a beacon of satellite or the emission from cosmic maser sources is received. The narrow-band spectral profile of the received signal leads us to compress the data up to the size less than the capacity of transmission rate on conventional telephone lines, that is to say a few tens of kbps. Additional devices for the real-time operation are A-D converters, PCs and modems, which are not special ones. As the link between the VLBI stations is performed by the conventional way, the system is applied not only to domestic VLBI networks, but also to international VLBI networks, in which a part of the VLBI stations is located in rural place. It has been confirmed that the cross-correlation is clearly detected for the compressed beacon wave data.

Third-generation (3G) wireless networks, based on a hierarchical cellular structure, support tiered levels of multimedia services. These services can be categorized as real-time and delay-sensitive, or non-real-time and delay- insensitive. Each call carries demand for one or more services in parallel; each with a guaranteed quality of service (QoS). Roaming is handled by handoff procedures between base stations (BSs) and the mobile subscribers (MSs) within the network. Metrics such as the probabilities of handoff failure, dropped calls and blocked calls; handoff transition time; and handoff rate are used to evaluate the handoff schemes, which also directly affects QoS. Previous researchers have proposed a fuzzy logic system (FLS) with neural encoding of the rule base and probabilistic neural network to solve the handoff decision as a pattern recognition problem in the set of MS signal measurements and mobility amid fading path uncertainties. Both neural approaches evalute only voice traffic in a closed, single- layer network of uniform cells. This paper proposed a new topology-preserving, self-organizing neural network (SONN) for both handoff and admission control as part of an overall resource allocation (RA) problem to support QoS in a three- layer, wideband CDMA HCS with dynamic loading of multimedia services. MS profiles include simultaneous service requirements, which are mapped to a new set of variables, defined in terms of the network radio resources (RRs). Simulations of the new SONN-based algorithms under various operating scenarios of MS mobility, dynamic loading, active set size, and RR bounds, using published traffic models of 3G services, compare their performance with earlier approaches.© (2000) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

Originally designed for data communication, the Internet is being considered for real-time voice and video communication (in applications such as telephony, voice conferencing, videophone and video conferencing). In order to become a legitimate alternative to the traditional circuit-switched telephone network, the internet must guarantee toll quality for voice communication, which translates into stringent requirements on voice packets, in particular in terms of delay, loss rate and service availability. In this paper, we seek to assess voice delay in the Internet. In this respect, we use network simulation to derive the end-to-end delay incurred by voice traffic for various network scenarios (i.e. path lengths, link bandwidths) and loading conditions, using different encoding and packetization schemes. As a consequence of the tradeoffs revealed by the delay results obtained, we propose a simple mechanism for end-to-end bandwidth management and admission control for voice streams.

The high data rates required to properly support emerging multimedia internet applications far exceed the capabilities of voiceband modems. For example, seamless real-time delivery of digital video clips might require connections up to 100 times faster. Asymmetric digital subscriber line (ADSL) modems provide residential internet users with a much-needed solution to this remote access bandwidth shortage. ADSL modems connected directly to the internet are installed in the telephone company central office, allowing remote access over the copper twisted pair telephone line by remote transceivers in the homes of residential customers. Because the internet data does not have to pass through the telephone switching network, the ADSL link can provide data rates in the Mbit/sec range. However, for ADSL modems to gain broad acceptance an displace their low rate voiceband counterparts, low cost, ease of installation, and high quality of service must be achieved.

IP telephony, a new technology to provide voice communication over traditional data networks, has the potential to revolutionize telephone communication within the modern enterprise. This innovation uses packetization techniques to carry voice conversations over IP networks. This packet switched technology promises new integrated services, and lower cost long-distance communication compared to traditional circuit switched telephone networks. Future satellites will need to carry IP traffic efficiently in order to stay competitive in servicing the global data- networking and global telephony infrastructure. However, the effects of Voice over IP over switched satellite channels have not been investigated in detail. To fully understand the effects of satellite channels on Voice over IP quality; several experiments were conducted at Lockheed Martin Telecommunications' Satellite Integration Lab. The result of those experiments along with suggested improvements for voice communication over satellite are presented in this document. First, a detailed introduction of IP telephony as a suitable technology for voice communication over future satellites is presented. This is followed by procedures for the experiments, along with results and strategies. In conclusion we hope that these capability demonstrations will alleviate any uncertainty regarding the applicability of this technology to satellite networks.© (1999) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

The presentation describes a new scenario for how to transform the existing telephone access network into a real broadband network. The objective is to introduce a low-cost network evolution scenario, which makes maximal use of the huge investments connected to this kind of networks. The existing network is sub-divided into a small cell network, each covering an area with an approximate radius of 1 kilometer. Inverse multiplexing of xDSL systems is used to aggregate bandwidth in the range of 100 to 1000 Mb/s between nodes in the network. VDSL modems able to carry about 25 - 50 Mb/s, are used on the last section from the node to the customer premises. These modems provide all households within each geographical cell with real broadband access to a large common bandwidth.

In this paper, an integrated monitoring architecture based on telephone maintenance networks for the optical CATV networks is proposed for the first time. Part of the field trial result is also shown. The integrated monitoring system includes the element monitoring sub-system (EMS), the RF performance monitoring sub-system (PMS), and the fiber monitoring sub-system (FMS). The EMS can monitor the active devices, including the optical transmitters and optical receivers in optical CATV networks. The PMS can monitor the RF performance of carrier-to-noise ratio (CNR), composite second order (CSO), composite triple beat (CTB), and per channel RF power level. The FMS can monitor the state of the fiber, and locate the fault for the point-to-point and the passive branching fiber networks. Both polling and alarm call-back methods can be executed. All the polling instructions and monitoring data are transmitted through the telephone maintenance networks. These three sub-systems share the common platform based on NT operation system, and user can access these three sub-systems just from a terminal. This integrated monitoring system not only provides the surveillance function, but also offers the administration function with user-friendly interface.

In Germany the Deutsche Telekom is realizing worldwide the largest installation of FITL systems under the programme OPAL (Optical Access Line) to build up a modem infrastructure in the eastern part of the country. At the end of 1997 about 1,5 million access lines will be based on fibre systems, mainly realized with FTTC solutions. Whilst in the first period of the installation PON structures have been favoured later on active networks have been implemented which showed much better economics and higher flexibility. The experiences with these systems are used to modernize also the existing telephone network in die western part in combination with a full digitization of all local exchanges. With the FITL systems the switching equipment can be concentrated on fewer locations leading to lower operation and maintenance costs. Deutsche Telekom operates besides the telephone network also worldwide the largest CATV network with about 25 million households passed. Currently work is going on to look for network upgrading techniques which allow the introduction of new broadband services like video-on-demand and broadband online services. FTTH is for all major network operators a long term target, however, due to the high digging costs and the still too high system prices FTTH is not yet economic and mainly the replacement of the first few hundred meters from the customers to the network node is much too costly. Fibre to the cabinet and new techniques like VDSL on copper lines or 64QAM on coax cables in connection with digital video transmission can be the next step to bring the fibre nearer to the home. Scenarios for these upgrading techniques will be presented and compared.

Residential broadband access network technology based on asynchronous transfer modem (ATM) will soon reach commercial availability. The capabilities provided by ATM access network promise integrated services bandwidth available in excess of those provided by traditional twisted pair copper wire public telephone networks. ATM to the side of the home placed need quality of service capability closest to the subscriber allowing immediate support for Internet services and traditional voice telephony. Other services such as desktop video teleconferencing and enhanced server-based application support can be added as part of future evolution of the network. Additionally, advanced subscriber home networks can be supported easily. This paper presents an updated summary of the standardization efforts for the ATM over HFC definition work currently taking place in the ATM forum's residential broadband working group and the standards progress in the IEEE 802.14 cable TV media access control and physical protocol working group. This update is fundamental for establishing the foundation for delivering ATM-based integrated services via a cable TV network. An economic model for deploying multi-tiered services is presenting showing that a single-tier service is insufficient for a viable cable operator business. Finally, the use of an ATM based system lends itself well to various deployment scenarios of synchronous optical networks (SONET).© (1996) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

A broad array of applications in the Public Switched Telephone Network (PSTN) require detailed information about type of call being carried. This information can be used to enhance service, diagnose transmission impairments, and increase available call capacity. The increase in data rates of modems and the increased usage of speech compression in the PSTN has rendered existing detection algorithms obsolete. Wavelets, specifically the Discrete Wavelet Transform (DWT), are a relatively new analysis tool in Digital Signal Processing. The DWT has been applied to signal processing problems ranging from speech compression to astrophysics. In this paper, we present a wavelet-based method of categorizing telephony traffic by call type. Calls are categorized as Voice or Data. Data calls, primarily modem and fax transmissions, are further divided by the International Telecommunications Union-Telephony (ITU-T), formerly CCITT, V-series designations (V.22bis, V.32, V.32bis, and V.34).

This paper describes a visualization tool which is a component of the NYNEX Network Exploratorium, a decision support platform for telephone interoffice network planning. The visualization tool provides a number of visualization styles suited to knowledge discovery in the network planning domain, and is designed for expansion to support additional styles. Visualizations are specific using a declarative language, which binds visual attributes directly to arbitrary attributes of objects in an object-oriented database. An overview of the language and some application examples are presented.© (1996) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

This paper reviews image telecommunications standards that exist or are under development. Included are a broad range of services, from facsimile to desktop videoconferencing systems and other multimedia applications. Image compression standards including T.4, T.6, JPEG, JBIG, MPEG, H.261 and H.263, and supporting communication protocol standards are examined. Communication facilities that carry these applications include the switched telephone network (PSTN), Integrated Services Data Network (ISDN), LAN and the mobile cellular telephone system. Still image communication of bilevel, gray scale and color images by means of Group 3 and Group 4 facsimile using T.4, T.6, JPEG and JBIG coding is included. Families of standards for videoconferencing applications (H.320 for N-ISDN, H.324 for PSTN, H.310/H.321 for ATM networks, H.322/H.323 for LAN and the T. 120 multi-point data protocol) are examined. Audio standards that support the videoconferencing applications are reviewed.

A broadband communication infrastructure (over 150 megabits per second), deployed almost everywhere outside the third world within 20 years, is a common planning assumption of governments, communication carriers, and information providers. The "structure" of this infrastructure has been variously projected as being that of the telephone network, the cable system, or the Internet. An argument is made that the telephone model, with features borrowed from the other two, will prevail. This model is used to project broad features of printing, publishing, and advertising. In support of this projection, printing is modeled purposefully, a document is printed to either archive it, give it to someone else, or use it (read, mark up, take along, etc.). In the broadband future, only the last is sustainable. Publishing is modeled as a four-stage chain of commerce from creator to buyer. The progress of both the document and its chain of payments is considered today and in the broadband scenario. Finally, advertising today and tomorrow is modeled as a 2x2x2 cube. One dimension contrasts the "notify/inform" and "persuade" aspects of advertising; another contrasts the consumer's role as passive recipient vs. active controller of what s/he hears and sees; the third views the institution of advertising as reflecting or setting societal values.

Hardware costs for telecommunications circuits can account for a maj or portion of total costs in Third World telem1icine progranis based on voice comunications. This is due to the relatively high cost associated with developing voice based coirimunications links to remote and isolated areas which are not already equipped with them. Iveloping satellite links specifically for telemedicine can be particularly expensive . However, more arid more remote and isolated areas in the Third World are already being sewed today by modern thin-route telecommunications systems . This is being done by exbending line of sight radio systems into remote areas using the so called point-to-multipoint microwave radio tecbnology. Point-to--multipoint radio systems have already been installed in several African and South American countries through a variety of financing arrangements such as funding by international agencies as well as funding by the countries theitselves. These systems reach into areas where health care seivices are often marginal. Health care workers — es, nurses , mid—wives and doctors , many, if not the majority of whom are women - often lack the support system that would provide for skills upgrading to allow them to benefit from the latest medical knowledge and advances. They also often lack the means to keep in touch with what is going on in the health field in other parts of their own country. By capitalizing on existing telecommunications links into such areas, telemedicine programs can be inlemented at costs far below those requiring the building of new networks for the purpose. This paper explores the possibilities of using existing telephone networks for implementing telemedicine projects in remote areas of the third world.

The project Mobimed started in 1988. The purpose of the project was to develop a telemedicine system where patient information text as well as data reflecting the status of various physiological parameters could be transmitted from a mobile terminal to a resident terminal at a hospital or similar place. The information transmitted was to be so comprehensive that a physician at the hospital together with the personnel attending the patient could make a diagnosis and suggest therapy. At the hospital all transmitted data should be stored automatically. Since September 1989 a prototype system is running in Sweden with one ambulance connected to two hospitals. This system is using the Mobitex* and speech transmission system for data transmission. As a first step the project is oriented towards ECG-monitoring. Mobimed has also been developed for laptop computers using the ordinary telephone network. In this case Mobimed can be used at home or anyplace where wireless transmission is not required. 1.© (1990) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

A very broad spectrum of technical literature has reported the activities and achievements in the field of fibre optics over the past ten years. The output in the past two years has been immense. Most of the reporting has been related to the world-wide achievements in optical transmission of data in public telephone networks. However, the less spectacular progress in commerce/industry has received significant exposure in the same period. The classical properties of optical fibres are being married to electronic circuitry to provide precision mark sensing in high speed processing equipment.© (1984) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

Integrated Telecom and Data Ring NetworkbyJens MaaloeNKT ElektronikDenmarkINTRODUCTIONThe NKT Tele- Ringnet is a combined digital telephone and data network based on a fibreoptical ring topology. The purpose of NKT Tele- Ringnet is to carry telephone and data com-munication as a digital PABX does. However, the optical system is not based on individuallines to a central mother exchange. It is a ringnet with an 8 Mbit /s transmission rateinterconnecting simple telephone concentrators. The advantage is that all the telephoneand data traffic can be multidropped over a large campus. In addition the signals arecarried by an optical dual fibre cable which has many advantages compared with conven-tional cobber cables, i.e. low attenuation, large noise immunity, no electromagnetic ra-diation, small size and low weight.NETWORK CONFIGURATIONThe basic requirement to the design of the integrated telecom and data network is to op-timize the system for application in digital environments, i.e. connection to digitalPABX's at the exchange side and connection to digital subscriber terminals in the localnetwork. In addition, it is also possible to terminate ordinary analog telephone subsets.In order to interconnect the system with the standard digital telecommunication system oftoday, the choice of architecture strictly follows the CCITT recommendations.The telephone and data network comprises three different units:telephone concentratorsa loop controllera service centreThe telephone concentrators and the loop controller are interconnected by optical fibresin a daisy chain topology. The logic topology of the network forms a ring. A blockdiagramindicating the topology of the telephone network is shown in fig. 1. The data frames trans-mitted on the fibres are in full accordance with the CCITT recommendations.The network elements are explained in details below.TELEPHONE CONCENTRATORThe telephone concentrator is a simple switch to which ordinary analog telephone subsetsand digital data equipment can be connected.

With the anticipated increase in demand for broadband services, and due to significant reductions in the cost of single mode fiber technology, it is likely that the local loop will make greater use of fiber optic technology. This will result in a tremendous increase in the required capabilities of a telephone network to handle the massive amounts of subscriber traffic.

This paper examines possible use of coherent based fiber optic systems in the telephone network. Coherent technology can increase the bandwidth of fiber based systems beyond the limits of current direct detection methods used for today's fiber optic systems by increasing the sensitivity of the receiver. Coherent technology can be used for one operating wavelength or in combination with Wavelength Division Multiplexing (WDM) devices for many operating wavelengths. "Wavelength management" will be necessary for the telephone companies should WDM devices be used with coherent technology. The most likely application for coherent systems is as transport for broadband ISDN when Gigabits per second (Gbps) bit rates and above are needed. High speed transport may be necessary between the remote electronics and broadband ISDN central office, between high speed fiber based Local Area Networks (LANs), and between central offices of a telephone company. System design should hinge on the standards developed for broadband ISDN. First, background information is given to identify the technology basics and application information, then telephone company operations issues are discussed, and finally conclusions are given.© (1987) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

This paper describes a low cost system that simul-taneously provides a data and a voice channel over a standard general service telephone network (GSTN) 2-wire connection. Time domain harmonic scaling (TDHS) [1] is used to frequency compress the voice signal. Data rates up to 1200 bits/s full-duplex have been achieved using a low complexity quadrature amplitude modulation (QAM) algorithm. A real-time implementation has been built using three TMS32010 DSP chips for full-duplex operation.

Bandwidth reduction for the transmission of sign language over telephone linesMansouria BoubekkerTufts University/New England Medical Center750 Washington Street, Boston, Massachusetts 02111ABSTRACTThis paper describes a system for the transmission of American Sign Language (ASL) over standard telephone lines. Videoquality digital images require a transmission bandwidth at least 4000 times wider than that allowable by the standard voicetelephone network. This study presents compression, segmentation and coding techniques that reduce the information contentof visual images while preserving the meaning of the transmitted message.The resulting processed information can be transmitted at digital rates within the range of the commercially available 9600baud modems. The simulated real -time animations of the decoded cartoon images used for the experimental data representASL and finger spelling with an intelligibility approximating 85 %. This result is within the range of the intelligibilityprovided by speech communication over telephone lines.1. INTRODUCTIONThe deaf individual is vastly isolated not only from the rest of the society but also from other members of the deafcommunity because of the difficulty to communicate at a distance. He or she can not benefit from the telephone network,widely and extensively used by the hearing society. Even special telephones with amplifiers, available to the hard of hearing,can not be used by profoundly deaf individuals. The primary form of communication used among individuals of the deafcommunity is American Sign Language. ASL is a visual /gestural language comparable to spoken languages in the sense thatit has its own structure, morphology and syntax[1,2,3]. However, it is an independent language that uses movements of thehands and arms as well as facial expressions instead of sounds.While different means of communication at a distance can be used by deaf individuals, they all present some majordrawbacks:- High resolution video telecommunication systems, the ideal means of communication for the transmission ofvisual images, are limited by the high video bandwidth requirement[4].- Video conferencing, provides a high quality means of communication for the deaf but its use is limited because ofits incompatibility with the actual public switched telephone network and its restriction to the immediate institu-tion[5].- Teletypewriters (TTY), the most common means of communication at a distance, for the deaf individual, limit thesystem by the slow rate of typed messages and the unnatural use of English as a mean of communication.An ideal telecommunication system for the deaf would be one that allows the transmission of visual images through the useof the switched telephone network. The transmission of visual images, rather than auditory signals using the standardtelephone system is difficult because of the large bandwidth requirement of a video quality image. However, digitaltransmission, through the use of commonly available modems, can be accomplished over audio telephone lines at a datarate of 9600 bits /s. A compression factor of at least 4000 is required for the transmission of visual images over digitalchannels.