通信类外文文献翻译蜂窝网络中的全双工通信设备

通信技术中的全双工通信的原理和实际应用全双工通信是一种在通信系统中允许数据同时进行双向传输的技术。

它允许发送方和接收方可以同时发送和接收信息，这样能够提高通信效率和传输速度。

本文将介绍全双工通信的原理和实际应用。

首先，让我们来了解全双工通信的原理。

全双工通信是通过使用两个独立的信道来实现的，一个用于发送数据，另一个用于接收数据。

这两个信道在物理上是分离的，因此可以同时进行数据的发送和接收。

通常，全双工通信中使用的信道可以是光纤、电缆、无线电波等。

实现全双工通信的一个重要组成部分是双工器。

双工器是一种用于在同一信道上实现双向通信的设备。

它可以分离发送和接收信号，使它们能够同时进行。

双工器可以将发送方的信号分离出来，然后将其传输到接收方，同时将接收方的信号分离出来，然后将其传输到发送方。

这样，发送方和接收方就可以在同一信道上同时进行通信。

全双工通信的实际应用非常广泛。

在电话系统中，全双工通信被广泛应用。

例如，当两个人进行电话对话时，每个人都可以同时说话和听对方说话，而不需要等待对方完成。

这样可以实现更加流畅和自然的对话。

另一个实际应用是在计算机网络中。

在局域网、广域网和互联网中，全双工通信被广泛用于数据的传输。

例如，在一个局域网中，当一个计算机发送数据给另一个计算机时，它可以同时接收来自另一个计算机的数据，而不需要等待。

这样可以提高数据传输的效率和速度。

此外，全双工通信还在无线通信中得到了广泛应用。

例如，在无线电对讲机中，全双工通信允许用户同时发送和接收语音信息。

这样可以实现快速和高效的信息交流。

总结一下，全双工通信是一种能够同时进行双向数据传输的通信技术。

它通过使用两个独立的信道和双工器来实现。

全双工通信的实际应用非常广泛，在电话系统、计算机网络和无线通信中都有所应用。

通过使用全双工通信，可以提高通信的效率和传输速度，实现更加流畅和高效的数据传输。

Wireless Communications＊byJoshua S。

Gans，Stephen P。

King and Julian Wright1. IntroductionIn 1895, Guglielmo Marconi opened the way for modern wireless communications by transmitting the three—dot Morse code for the letter ‘S’ over a distance of th ree kilometers using electromagnetic waves。

From this beginning，wireless communications has developed into a key element of modern society. From satellite transmission, radio and television broadcasting to the now ubiquitous mobile telephone，wireless communications has revolutionized the way societies function.This chapter surveys the economics literature on wireless communications。

Wireless communications and the economic goods and services that utilise it have some special characteristics that have motivated specialised studies。

First, wireless communications relies on a scarce resource –namely，radio spectrum –the property rights for which were traditionally vested with the state. In order to foster the development of wireless communications (including telephony and broadcasting）those assets were privatised。

Triple wireless voice data transmission system designStudent :,Instructor :, UniversityEvery day, in our work and in our leisure time, we come in contact with and use a variety of modern communication media, the most common being the telephone, radio, television, and the Internet. Though these media we are able to communicate (nearly) instantaneously with people on diffident continents, transact our daily business, and receive information about various developments and events of note that occur all around the world. Electronic mail and facsimile transmission have made it possible to rapidly communicate written message across great distances.Wireless communications. The development of wireless communications stems from the works of Oersted, Faraday, Gauss, Maxwell, and Hertz. In1820, Oersted demonstrated that an electric current produces a magnetic field. On August 29,1831,Michael Faraday showed that an induced current is produced by moving a magnet in the vicinity of a conductor. Thus, he demonstrated that a changing magnetic field produces an electric field. With this early work as background, James C. Maxwell in 1864 predicted the existence of electromagnetic radiation and formulated the basic theory that has been in use for over a century. Maxwell’s theory was verified experimentally by Hertz in 1887.In 1894, a sensitive device that could device that could detect radio signals, called the coherer, was used by its inventor Oliver Lodge to demonstrate wireless communication over a distance of 150 yards at Oxford, England. Guglielmo Marconi is credited with the development of wireless telegraphy. Marconi demonstrated the transmission of radio signals at a distance of approximately 2 kilometers in 1895. Two years later, in 1897 , he patented a radio telegraph system and established the Wireless Telegraph and Signal Company. On December 12, 1901, Marconi received a radio signal at Signal Hill in Newfoundland, which was transmitted from Cornwall, England, a distance of about 1700 miles.The invention of the vacuum tube was especially instrumental in the development of radio communication system .The vacuum diode was invented by Fleming in 1904 and the vacuum triode amplifier was invented by De Forest in 1906, as previously indicated. The invention of the triode made radio broadcast possible in the early part of the twentieth century. Amplitude modulation (AM) broadcast was initiated in 1920 when radio station KDKA, Pittsburgh, went on the air. From that date, AM radio broadcasting grew rapidly across the country and around the world. The super heterodyne AM radio receiver, as we know it today, was invented by Edwin Armstrong during World War I. Another significant development in radio communications was the invention of Frequency modulation (FM), also by Armstrong.In 1933, Armstrong built and demonstrated the first FM communication system. However, the use of FM was slow to develop compared with AM broadcast. It was not until the end of World War II that FM broadcast gained in popularity and developed commercially.The first television system was built in the United States by V. K. Zworykin and demonstrated in 1929. Commercial television broadcasting began in London in 1936 by the British Broadcasting Corporation(BBC) . Five years later the Federal Communications Commission(FCC) authorized television broadcasting in the United States.ELEMENTS OF AN ELECTRICAL COMMUNICA SYSTEM Electrical communication systems are designed to send messages or information from a source that generates the message to one more destinations. In general, a communication system can be represented by the functional block diagram shown . The information generated by the source may be of the form of voice (speech source), a picture (image source), or plain text in some particular language, such as English , Japanese, German , French, etc. An essential feature of any source that generates information is that its output is described in probabilistic terms; i.e., the output of a source is not deterministic. Otherwise, there would be no need to transmit the message.A transducer is usually required to convert the output of a source into an electrical signal that is suitable for transmission. For example, a microphone serves as the transducer that converts an acoustic speech signal. At the destination, a similar transducer is required to convert the electrical signals that are received into a form that is suitable for the user; e.g., acoustic signals, images, etc.The heart of the communication system consists of three basic parts, namely, the transmitter, the channel, and the receiver. The functions performed by these three elements are described next.The Transmitter. The Transmitter converts the electrical signal into a form that is suitable for transmission though the physical channel or transmission medium. For example, in radio and TV broadcast, the Federal Communications Commission (FCC) specifies the frequency range for each transmitting station. Hence, the transmitter must translate the information signal to be transmitted into the appropriate The Transmitter range that matches the frequency allocation assigned to the transmitter. Thus, signal transmitted by multiple radio station do not interfere with one another. Similar functions are performed in telephone communication systems where the electrical speech signals from many users are transmitted over the same wire.In general, the transmitter performs the matching of the message signal to the channel by a process called modulation. Usually, modulation involves the use of the information signal to systematically vary either the amplitude, frequency, or phase of a sinusoidal carrier. For example, in AM radio broadcast, the information signal that is transmitted is contained in the amplitude variations of the sinusoidal carrier, which is the center frequency in the amplitude modulation. In FM radio broadcast., the information signal that is transmitted is contained in the frequency variations of thesinusoidal carrier. This is an example of frequency modulation. Phase modulation (PM) is yet a third method for impressing the information signal on a sinusoidal carrier.In general, carrier modulation such as AM, FM, and PM is performed at the transmitter, as indicated above, to convert the information signal to a form that matches the characteristics of the channel. Thus, though the process of modulation, the choice of the type of modulated in frequency to match the allocation of the channel. The choice of the type of modulation is based on several factors, such as the amount of bandwidth over the channel, the type of noise and the interference that the signal encounters in transmission. In any case, the modulation process makes it possible to accommodate the transmission of multiple messages from many users over the same physical channel.In addition to modulation, other functions that are usually performed at the transmitter are filtering of the information-bearing signal , amplification of the modulated signal, and in case of wireless transmission, radiation of the signal by means of a transmitting antenna.The channel. The communications channel is the physical medium that is used to send the signal from the transmitter to the receiver. In wireless transmission, the channel is usually the atmosphere (free space). On the other hand, telephone channels usually employ a variety of physical media, including wirelines, optical fiber cables, and wireless (microwave radio). Whatever the physical medium for signal transmission, the essential feature is that the transmitted signal is corrupted in a random manner by a variety of possible mechanisms. The most common from of signal degradation comes in the form of additive noise ,which is generated at the front end of the receiver, where signal amplification is performed. This noise is often called thermal noise. In wire less transmission, additional additive disturbances are man-made noise, and atmospheric noise picked up by a receiving antenna. Automovile ignition noise is an example of man-made noise, and electrical lightning discharges from thunderstorms is an example of atmospheric noise. Interference from other users of the channel is another form of additive noise that often arises in both wireless and wire line communication systems .In some radio communication channels, such as the ionospheric channel that is used for long range ,short-wave radio transmission, another form of signal degradation is multipath propagation. Such signal distortion is characterized as a nonadditive signal disturbance which manifests itself as time variations in the signal amplitude, usually called fading .Both additive and nonadditive signal distortions are usually characterized as random phenomena and described in statistical terms. The effect of these signal distortions must be taken into account on the design of the communication system.In the design of a communication system, the system, the system designer works with mathematical models that statistically characterize he signal distortion encountered on physical channels. Often, the statistical description that is used in mathematical model is a result of actual empirical measurements obtained from experiments involving signal transmission over such channels .In such cases , there isa physical justification for the mathematical model used in the design of communication systems. On the other hand, in some communication system designs ,the statistical characteristics of the channel may vary significantly with time. In such cases, the system design may designer may design a communication system that is robust to the variety of signal distortions. This can be accomplished by having the system adapt some of its parameters to the channel distortion encountered.The receiver. The function of the receiver is to recover the message signal contained in the received signal. If the message signal is transmitted by carrier modulation, the receiver performs carrier demodulation in order to extract the message from the sinusoidal carrier. Since the signal demodulation is performed in the presence of additive noise and possibly other signal distortion, the demodulated message signal is generally degraded to some extent by the presence of these distortions in the received signal. As we shall see, the fidelity of the additive noise, the type and strength of any other additive interference, and the type of any nonadditive interference.Besides performing the primary function of signal demodulation, the receiver also performs a number of peripheral functions, including signal filtering and noise suppression.Digital Communication SystemAn electrical communication system in rather broad terms based on the implicit assumption that message signal is a continuous timevarying waveform. We refer to such continuous-time signal waveforms as analog sources. Analog signal can be transmitted directly via modulation over the communication channel and demodulated accordingly at the receiver. We call such a i communication system an analog communication system.Alternatively, an analog source output may be converted into a digital form and the message can be transmitted via digital modulation as a digital signal at the receiver. There are some potential advantage to transmitting an analog signal by means of digital modulation. The most important reason is that signal fidelity is better controlled though digital transmission than analog transmission. In particular, digital transmission allows us to regenerate the digital signal in long-distance transmission, thus eliminating effects of noise at each regeneration point. In contrast, the noise added in analog transmission is amplified along with the signal when amplifiers are used periodically to boost the signal level in long-distance transmission. Another reason for choosing digital transmission over analog is that the analog message signal may be highly redundant. With digital processing, redundancy may be removed prior to modulation, thus conserving channel bandwidth. Yet a third reason may be that digital communication systems are often cheaper to implement.In some applications, the information to be transmitted is inherently digital; e.g., in the form of English text, computer data, etc. In such cases, the information source that generates the data is called a discrete (digital)source.In a digital communication systems , the some applications, the functional operations performed at the transmitter and receiver must be expanded to includemessage signal discrimination at the transmitter and message signal synthesis or interpolation at the receiver. Additional functions include redundancy removal, and channel coding and decoding.The source output may be either an analog signal, such as audio or video signal, or a digital signal , such as the output of a computer which is discrete in time and has a finite number of output characters. In a digital communication system, the message produced by the source are usually converted into a sequence of binary digits as possible. In other words, we seek inefficient representation of the source output of either an analog or a digital source into a sequence of binary digits is called source encoding or date compression.The sequence of binary digits from the coerce encoder, which we call the information sequence is passed to the channel encoder. The purpose of the channel encoder is to introduce, in a controlled manner, some redundancy in binary information sequence which can be used at the receiver to overcome the effects of noise and interference encountered in the transmission of the signal though the channel. Thus the added redundancy serves to increase the reliability of the received data and improves the fidelity in decoding the deceived signal. In fact, redundancy serves in the information sequence aids the receiver in decoding the desired information sequence .The binary sequence at the output of the channel encoder is passed to the digital modulator, which servers as the interface to the communications channel. Since nearly all of the communication channels encountered in practice are capable of transmitting electrical signals (waveforms), the primary purpose of the digital modulator is to map the binary information sequence into signal waveforms.At the receiving end of a digital communication system, the digital demodulator processes the channel-corrupted transmitted waveform and reduces reduce each waveform to a signal number that represents an estimate of the transmitted data symbol (binary or Mary) . When there is no redundancy in the transmitted information, the demodulator must decide which of the M waveform was transmitted in any given time interval. A measure of how well the demodulator and encoder perform is the frequency with which errors occur in the decoded sequence.As a final step, when an analog output is desired, the source decoder accepts the output sequence from the channel and , from knowledge of the source-encoding method used, attempts to reconstruct the original signal from the source.双工无线语音数据传输系统的设计学生：学院指导老师：汉大学在日常的工作和生活中，人们每天都要接触和使用大量的现代通信系统和通信媒介，其中最常见的是电话，无线电广播，电视和因特网。

Multi-Code TDMA (MC-TDMA) for Multimedia Satellite Communications用于多媒体卫星通信的MC--TDMA（多码时分多址复用）R. Di Girolamo and T. Le-NgocDepartment ofa Electricl and Computer Engineering - Concordia University1455 de Maisonneuve Blvd. West, Montreal, Quebec, Canada, H3G 1M8 ABSTRACT摘要In this paper, we propose a multiple access scheme basedon a hybrid combination of TDMA and CDMA,在这篇文章中，我们提出一种基于把时分多址复用和码分多址复用集合的多址接入方案。

referred toas multi-code TDMA (MC-TDMA). 称作多码—时分多址复用The underlying TDMAframe structure allows for the transmission of variable bitrate (VBR) information,以TDMA技术为基础的帧结构允许传输可变比特率的信息while the CDMA provides inherentstatistical multiplexing.和CDMA提供固有的统计特性多路复用技术The system is studied for a multimediasatellite environment with long-range dependentdata traffic,and VBR real-time voice and video traffic研究这个系统是为了在远程环境下依赖数据传输和可变比特率的语音和视频传输的多媒体卫星通信系统 . Simulationresults show that with MC-TDMA, the data packetdelay and the probability of real-time packet loss can bemaintained low. 仿真结果表明：采用MC-TDMA的多媒体卫星通信，数据包延时和实时数据丢失的可能性可以保持很低。

附录一、英文原文：Detecting Anomaly Traffic using Flow Data in the realVoIP networkI. INTRODUCTIONRecently, many SIP[3]/RTP[4]-based VoIP applications and services have appeared and their penetration ratio is gradually increasing due to the free or cheap call charge and the easy subscription method. Thus, some of the subscribers to the PSTN service tend to change their home telephone services to VoIP products. For example, companies in Korea such as LG Dacom, Samsung Net- works, and KT have begun to deploy SIP/RTP-based VoIP services. It is reported that more than five million users have subscribed the commercial VoIP services and 50% of all the users are joined in 2009 in Korea [1]. According to IDC, it is expected that the number of VoIP users in US will increase to 27 millions in 2009 [2]. Hence, as the VoIP service becomes popular, it is not surprising that a lot of VoIP anomaly traffic has been already known [5]. So, Most commercial service such as VoIP services should provide essential security functions regarding privacy, authentication, integrity and non-repudiation for preventing malicious traffic. Particu- larly, most of current SIP/RTP-based VoIP services supply the minimal security function related with authentication. Though secure transport-layer protocols such as Transport Layer Security (TLS) [6] or Secure RTP (SRTP) [7] have been standardized, they have not been fully implemented anddeployed in current VoIP applications because of the overheads of implementation and performance. Thus, un-encrypted VoIP packets could be easily sniffed and forged, especially in wireless LANs. In spite of authentication,the authentication keys such as MD5 in the SIP header could be maliciously exploited, because SIP is a text-based protocol and unencrypted SIP packets are easily decoded. Therefore, VoIP services are very vulnerable to attacks exploiting SIP and RTP. We aim at proposing a VoIP anomaly traffic detection method using the flow-based traffic measurement archi-tecture. We consider three representative VoIP anomalies called CANCEL, BYE Denial of Service (DoS) and RTP flooding attacks in this paper, because we found that malicious users in wireless LAN could easily perform these attacks in the real VoIP network. For monitoring VoIP packets, we employ the IETF IP Flow Information eXport (IPFIX) [9] standard that is based on NetFlow v9. This traffic measurement method provides a flexible and extensible template structure for various protocols, which is useful for observing SIP/RTP flows [10]. In order to capture and export VoIP packets into IPFIX flows, we define two additional IPFIX templates for SIP and RTP flows. Furthermore, we add four IPFIX fields to observe packets which are necessary to detect VoIP source spoofing attacks in WLANs.II. RELATED WORK[8] proposed a flooding detection method by the Hellinger Distance (HD) concept. In [8], they have pre- sented INVITE, SYN and RTP flooding detection meth-ods. The HD is the difference value between a training data set and a testing data set. The training data set collected traffic over n sampling period of duration Δ testing data set collected traffic next the training data set in the same period. If the HD is close to ‘1’, this testing data set is regarded as anomaly traffic. For using this method, they assumed that initial training data set didnot have any anomaly traffic. Since this method was based on packet counts, it might not easily extended to detect other anomaly traffic except flooding. On the other hand, [11] has proposed a VoIP anomaly traffic detection method using Extended Finite State Machine (EFSM). [11] has suggested INVITE flooding, BYE DoS anomaly traffic and media spamming detection methods. However, the state machine required more memory because it had to maintain each flow. [13] has presented NetFlow-based VoIP anomaly detection methods for INVITE, REGIS-TER, RTP flooding, and REGISTER/INVITE scan. How-ever, the VoIP DoS attacks considered in this paper were not considered. In [14], an IDS approach to detect SIP anomalies was developed, but only simulation results are presented. For monitoring VoIP traffic, SIPFIX [10] has been proposed as an IPFIX extension. The key ideas of the SIPFIX are application-layer inspection and SDP analysis for carrying media session information. Yet, this paper presents only the possibility of applying SIPFIX to DoS anomaly traffic detection and prevention. We described the preliminary idea of detecting VoIP anomaly traffic in [15]. This paper elaborates BYE DoS anomaly traffic and RTP flooding anomaly traffic detec-tion method based on IPFIX. Based on [15], we have considered SIP and RTP anomaly traffic generated in wireless LAN. In this case, it is possible to generate the similiar anomaly traffic with normal VoIP traffic, because attackers can easily extract normal user information from unencrypted VoIP packets. In this paper, we have extended the idea with additional SIP detection methods using information of wireless LAN packets. Furthermore, we have shown the real experiment results at the commercial VoIP network.III. THE VOIP ANOMALY TRAFFIC DETECTION METHOD A. CANCEL DoS Anomaly Traffic DetectionAs the SIP INVITE message is not usually encrypted, attackers could extract fields necessary to reproduce the forged SIP CANCEL message by sniffing SIP INVITE packets, especially in wireless LANs. Thus, we cannot tell the difference between the normal SIP CANCEL message and the replicated one, because the faked CANCEL packet includes the normal fields inferred from the SIP INVITE message. The attacker will perform the SIP CANCEL DoS attack at the same wireless LAN, because the purpose of the SIP CANCEL attack is to prevent the normal call estab-lishment when a victim is waiting for calls. Therefore, as soon as the attacker catches a call invitation message for a victim, it will send a SIP CANCEL message, which makes the call establishment failed. We have generated faked SIP CANCEL message using sniffed a SIP INVITE in SIP header of this CANCEL message is the same as normal SIP CANCEL message, because the attacker can obtain the SIP header field from unencrypted normal SIP message in wireless LAN environment. Therefore it is impossible to detect the CANCEL DoS anomaly traffic using SIP headers, we use the different values of the wireless LAN frame. That is, the sequence number in the frame will tell the difference between a victim host and an attacker. We look into source MAC address and sequence number in the MAC frame including a SIP CANCEL message as shown in Algorithm 1. We compare the source MAC address of SIP CANCEL packets with that of the previously saved SIP INVITE flow. If the source MAC address of a SIP CANCEL flow is changed, it will be highly probable that the CANCEL packet is generated by a unknown user. However, the source MAC address could be spoofed. Regarding source spoofing detection, we employ the method in [12] that uses sequence numbers of frames. We calculate the gap between n-th and (n-1)-th frames. As the sequence number field in a MAC header uses 12 bits, it varies from 0 to 4095. When we find that the sequence number gap between a single SIP flow is greater than the threshold value of N that willbe set from the experiments, we determine that the SIP host address as been spoofed for the anomaly traffic.B. BYE DoS Anomaly Traffic DetectionIn commercial VoIP applications, SIP BYE messages use the same authentication field is included in the SIP IN-VITE message for security and accounting purposes. How-ever, attackers can reproduce BYE DoS packets through sniffing normal SIP INVITE packets in wireless faked SIP BYE message is same with the normal SIP BYE. Therefore, it is difficult to detect the BYE DoS anomaly traffic using only SIP header sniffing SIP INVITE message, the attacker at the same or different subnets could terminate the normal in- progress call, because it could succeed in generating a BYE message to the SIP proxy server. In the SIP BYE attack, it is difficult to distinguish from the normal call termination procedure. That is, we apply the timestamp of RTP traffic for detecting the SIP BYE attack. Generally, after normal call termination, the bi-directional RTP flow is terminated in a bref space of time. However, if the call termination procedure is anomaly, we can observe that a directional RTP media flow is still ongoing, whereas an attacked directional RTP flow is broken. Therefore, in order to detect the SIP BYE attack, we decide that we watch a directional RTP flow for a long time threshold of N sec after SIP BYE message. The threshold of N is also set from the 2 explains the procedure to detect BYE DoS anomal traffic using captured timestamp of the RTP packet. We maintain SIP session information between clients with INVITE and OK messages including the same Call-ID and 4-tuple (source/destination IP Address and port number) of the BYE packet. We set a time threshold value by adding Nsec to the timestamp value of the BYE message. The reason why we use the captured timestamp is that a few RTP packets are observed under second. If RTP traffic is observed after the time threshold, this willbe considered as a BYE DoS attack, because the VoIP session will be terminated with normal BYE messages. C. RTP Anomaly Traffic Detection Algorithm 3 describes an RTP flooding detection method that uses SSRC and sequence numbers of the RTP header. During a single RTP session, typically, the same SSRC value is maintained. If SSRC is changed, it is highly probable that anomaly has occurred. In addition, if there is a big sequence number gap between RTP packets, we determine that anomaly RTP traffic has happened. As inspecting every sequence number for a packet is difficult, we calculate the sequence number gap using the first, last, maximum and minimum sequence numbers. In the RTP header, the sequence number field uses 16 bits from 0 to 65535. When we observe a wide sequence number gap in our algorithm, we consider it as an RTP flooding attack.IV. PERFORMANCE EVALUATIONA. Experiment EnvironmentIn order to detect VoIP anomaly traffic, we established an experimental environment as figure 1. In this envi-ronment, we employed two VoIP phones with wireless LANs, one attacker, a wireless access router and an IPFIX flow collector. For the realistic performance evaluation, we directly used one of the working VoIP networks deployed in Korea where an 11-digit telephone number (070-XXXX-XXXX) has been assigned to a SIP wireless SIP phones supporting , we could make calls to/from the PSTN or cellular phones. In the wireless access router, we used two wireless LAN cards- one is to support the AP service, and the other is to monitor packets. Moreover, in order to observe VoIP packets in the wireless access router, we modified nProbe [16], that is an open IPFIX flow generator, to create and export IPFIX flows related with SIP, RTP, and information. As the IPFIX collector, we have modified libipfix so that it could provide the IPFIX flow decoding function for SIP, RTP, and templates. We used MySQL for the flow DB.B. Experimental ResultsIn order to evaluate our proposed algorithms, we gen-erated 1,946 VoIP calls with two commercial SIP phones and a VoIP anomaly traffic generator. Table I showsour experimental results with precision, recall, and F-score that is the harmonic mean of precision and recall. In CANCEL DoS anomaly traffic detection, our algorithm represented a few false negative cases, which was related with the gap threshold of the sequence number in MAC header. The average of the F-score value for detecting the SIP CANCEL anomaly is %.For BYE anomaly tests, we generated 755 BYE mes-sages including 118 BYE DoS anomalies in the exper-iment. The proposed BYE DoS anomaly traffic detec-tion algorithm found 112 anomalies with the F-score of %. If an RTP flow is terminated before the threshold, we regard the anomaly flow as a normal one. In this algorithm, we extract RTP session information from INVITE and OK or session description messages using the same Call-ID of BYE message. It is possible not to capture those packet, resulting in a few false-negative cases. The RTP flooding anomaly traffic detection experiment for 810 RTP sessions resulted in the F score of 98%.The reason of false-positive cases was related with the sequence number in RTP header. If the sequence number of anomaly traffic is overlapped with the range of the normal traffic, our algorithm will consider it as normal traffic.V. CONCLUSIONSWe have proposed a flow-based anomaly traffic detec-tion method against SIP and RTP-based anomaly traffic in this paper. We presented VoIP anomaly traffic detection methods with flow data on the wireless access router. We used the IETF IPFIX standard to monitor SIP/RTP flows passing through wireless access routers, because its template architecture is easily extensible to several protocols. For this purpose, we defined two new IPFIX templates for SIP and RTP traffic and four new IPFIX fields for traffic. Using these IPFIX flow templates,we proposed CANCEL/BYE DoS and RTP flooding traffic detection algorithms. From experimental results on the working VoIP network in Korea, we showed that our method is able to detect three representative VoIP attacks on SIP phones. In CANCEL/BYE DoS anomaly trafficdetection method, we employed threshold values about time and sequence number gap for classfication of normal and abnormal VoIP packets. This paper has not been mentioned the test result about suitable threshold values. For the future work, we will show the experimental result about evaluation of the threshold values for our detection method.二、英文翻译：交通流数据检测异常在真实的世界中使用的VoIP网络一 .介绍最近,许多SIP[3],[4]基于服务器的VoIP应用和服务出现了,并逐渐增加他们的穿透比及由于自由和廉价的通话费且极易订阅的方法。

5G无线通信网络中英文对照外文翻译文献(文档含英文原文和中文翻译)翻译：5G无线通信网络的蜂窝结构和关键技术摘要第四代无线通信系统已经或者即将在许多国家部署。

然而，随着无线移动设备和服务的激增，仍然有一些挑战尤其是4G所不能容纳的，例如像频谱危机和高能量消耗。

无线系统设计师们面临着满足新型无线应用对高数据速率和机动性要求的持续性增长的需求，因此他们已经开始研究被期望于2020年后就能部署的第五代无线系统。

在这篇文章里面，我们提出一个有内门和外门情景之分的潜在的蜂窝结构，并且讨论了多种可行性关于5G无线通信系统的技术，比如大量的MIMO技术，节能通信，认知的广播网络和可见光通信。

面临潜在技术的未知挑战也被讨论了。

介绍信息通信技术（ICT）创新合理的使用对世界经济的提高变得越来越重要。

无线通信网络在全球ICT战略中也许是最挑剔的元素，并且支撑着很多其他的行业，它是世界上成长最快最有活力的行业之一。

欧洲移动天文台（EMO）报道2010年移动通信业总计税收1740亿欧元,从而超过了航空航天业和制药业。

无线技术的发展大大提高了人们在商业运作和社交功能方面通信和生活的能力无线移动通信的显著成就表现在技术创新的快速步伐。

从1991年二代移动通信系统(2G)的初次登场到2001年三代系统（3G）的首次起飞，无线移动网络已经实现了从一个纯粹的技术系统到一个能承载大量多媒体内容网络的转变。

4G无线系统被设计出来用来满足IMT-A技术使用IP面向所有服务的需求。

在4G系统中，先进的无线接口被用于正交频分复用技术（OFDM），多输入多输出系统（MIMO）和链路自适应技术。

4G无线网络可支持数据速率可达1Gb/s的低流度，比如流动局域无线访问，还有速率高达100M/s的高流速，例如像移动访问。

LTE系统和它的延伸系统LTE-A，作为实用的4G系统已经在全球于最近期或不久的将来部署。

然而，每年仍然有戏剧性增长数量的用户支持移动宽频带系统。

毕业设计（论文）的外文文献翻译原始资料的题目/来源：Fundamentals of wireless communications by David Tse翻译后的中文题目：无线通信基础专业通信工程学生王晓宇学号110240318班号1102403指导教师杨洪娟翻译日期2015年6月15日外文文献的中文翻译7．mimo：空间多路复用与信道建模本书我们已经看到多天线在无线通信中的几种不同应用。

在第3章中，多天线用于提供分集增益，增益无线链路的可靠性，并同时研究了接受分解和发射分解，而且，接受天线还能提供功率增益。

在第5章中，我们看到了如果发射机已知信道，那么多采用多幅发射天线通过发射波束成形还可以提供功率增益。

在第6章中，多副发射天线用于生产信道波动，满足机会通信技术的需要，改方案可以解释为机会波束成形，同时也能够提供功率增益。

章以及接下来的几章将研究一种利用多天线的新方法。

我们将会看到在合适的信道衰落条件下，同时采用多幅发射天线和多幅接收天线可以提供用于通信的额外的空间维数并产生自由度增益，利用这些额外的自由度可以将若干数据流在空间上多路复用至MIMO信道中，从而带来容量的增加：采用n副发射天线和接受天线的这类MIMO 信道的容量正比于n。

过去一度认为在基站采用多幅天线的多址接入系统允许若干个用户同时与基站通信，多幅天线可以实现不同用户信号的空间隔离。

20世纪90年代中期，研究人员发现采用多幅发射天线和接收天线的点对点信道也会出现类似的效应，即使当发射天线相距不远时也是如此。

只要散射环境足够丰富，使得接受天线能够将来自不同发射天线的信号分离开，该结论就成立。

我们已经了解到了机会通信技术如何利用信道衰落，本章还会看到信道衰落对通信有益的另一例子。

将机会通信与MIMO技术提供的性能增益的本质进行比较和对比是非常的有远见的。

机会通信技术主要提供功率增益，改功率增益在功率受限系统的低信噪比情况下相当明显，但在宽带受限系统的高信噪比情况下则很不明显。

一、汉译英1、时分多址：TDMA （Time Division Multiple Address/ Time Division Multiple Access）2、通用无线分组业务：GPRSGeneral Packet Radio Service3、国际电报电话咨询委员会：CCITT4、同步数字体系：SDH Synchronous Digital Hierarchy （同步数字序列）5、跳频扩频：FHSS frequency hopping spread spectrum6、同步转移模块：STM synchronous transfer module7、综合业务数字网：ISDNIntegrated Services Digital Network8、城域网：MAN Metropolitan Area Network9、传输控制协议/互联网协议：TCP/IPTransmission Control Protocol/Internet Protocol10、服务质量：QOS Quality of Service11、中继线：trunk line12、传输速率：transmission rate13、网络管理：network management14、帧结构：frame structure15、移动手机：Mobile Phone 手机 Handset16、蜂窝交换机：（Cellular switches）（电池开关cell switch）(cell 蜂房)17、天线：Antenna18、微处理器：microprocessor19、国际漫游：International roaming20、短消息：short message21、信噪比：SNR(Signal to Noise Ratio)22、数字通信：Digital communication23、系统容量：system capacity24、蜂窝网：cell network（cellular network）（Honeycomb nets）25、越区切换：Handover26、互联网：internet27、调制解调器：modem28、频谱：spectrum29、鼠标：Mouse30、电子邮件：electronic mail E-mail31、子网：subnet32、软件无线电：software defined radios33、网络资源：network resources 八、英译汉1、mobile communication：移动通信2、Computer user：计算机用户3、Frame format：帧格式4、WLAN：wireless local area network 无线局域网络5、Communication protocol：通信协议6、Transmission quality：传输质量7、Remote terminal：远程终端8、International standard：国际标准9、GSM：全球移动通信系统Global System for Mobile Communications10、CDMA：码分多址Code Division Multiple Access11、ITU：国际电信联盟International Telecommunication Union12、PCM：pulse code modulation 脉冲编码调制13、WDM：波分复用Wavelength Division Multiplex14、FCC：联邦通信委员会Federal communications commission15、PSTN：公用电话交换网Public Switched Telephone Network16、NNI：网络节点借口Network Node Interface17、WWW：万维网World Wide Web18、VOD：视频点播Video-On-Demand19、VLR：访问位置寄存器Visitor Location Register20、MSC：移动交换中心Mobile Switching Centre21、HLR：原籍位置寄存器Home Location Register22、VLSI：超大规模集成电路Very Large Scale Integrated Circuits23、Bluetooth technology：蓝牙技术24、Matched filter：匹配滤波器25、ADSL：非对称数字用户环路Asymmetrical Digital Subscriber Loop非对称数字用户线路(Asymmetric Digital Subscriber Line)26、GPS：全球定位系统Global Position System27、ATM：异步传输模式Asynchronous Transfer Mode1、脉冲编码调制（PCM）依赖于三个独立的操作：抽样、量化和编码。

中文翻译2．1理想蜂窝小区的覆盖范围根据标准传播模型SPM，各个地区针对不同的无线信号频段分别进行传播模型校正，得到较适合该地区该频段的传播模型：d表示接收机与基站间的距离；HT x为发射机高度；HRx为接收机高度；DiffLoss表示散射造成的路径损耗；f(clutter)表示各个clutter损耗的加权平均，表示为Kc。

通常情况下，一个区域内的移动终端接收电平在-85dBm以上，可以认为该区域有较良好的无线网络覆盖。

因此蜂窝基站小区覆盖边界处的接收电平为-85dBm、平均天线口发射功率51dBm来推算理论覆盖范围。

2．2 GSM理论网络容量与载干比C/I间的计算方法载干比C/I通常被用来衡量GSM网络的移动通话质量若使用MRP（多重频率复用方式）来进行频率规划的话，则理论计算规划可实现的C/I，可以应用以下的公式大致推算出：其中a表示传播模型中的路径损耗斜率；D表示相邻2个小区群中位置相对应的2个小区中心之间的距离，即最近同频小区距离，也称为频率复用距离；R 为小区六边形外接圆半径，即小区覆盖半径；Ki为传播条件、天线方向去耦、天线下倾等因素的综合修正因子（建议取值范围为-5~0）。

针对一个地区具体的网络a和Ki为相对固定的值。

2．3 GSM系统网络容量与无线通话质量间的关系GSM网络无线通话质量的主要衡量标准为载干比C/I，当一块区域的GSM网络载干比C/I>9dB时，可以认为该区域拥有较好无线通话质量。

从章节2.2中的公式中得到推论，在一个确定的无线环境中，蜂窝小区连续均匀覆盖，即小区的覆盖距离为定值，若载干比C/I值恒定，则GSM理论网络容量与可用载频数成正比。

若GSM理论网络容量恒定，则载干比C/I值与可用载频数成正比。

3．2 COMMON BCCH技术COMMON BCCH是双频共BTS的一种方式，利用DCS1800M的无线频率资源来补充GSM900M的单蜂窝扇区的载频数，从而实现容量上的增加。

蜂窝网络中全双工D2D通信功率控制赵季红;何强;曲桦;栾智荣【摘要】在蜂窝网络中,采用全双工传输的设备直通(D2D)通信可以共享蜂窝通信的信道资源,提升频谱利用率和系统吞吐量.针对单对全双工D2D用户复用单个蜂窝用户的上行信道资源时,用户之间会产生同频干扰的问题,提出了一种低复杂度的功率控制算法.该算法在保证全双工D2D用户和蜂窝用户(CU)的服务质量(QoS)的前提下,最大化全双工D2D链路的吞吐量.仿真结果表明,该算法能够提高全双工D2D链路的吞吐量;全双工D2D链路吞吐量取决于蜂窝用户的QoS要求、相对距离以及自干扰消除数量的限制.%Device-to-device (D2D) communications that based on the wireless full-duplex transmission mode can not only reuse the cellular users' spectrum resource,but also significantly improve the spectral efficiency and the system throughout in wireless cellular networks.However,full-duplex D2D communications may generate same frequency interference to the reused cellular user (CU) while full-duplexD2D users share the same resources as CU in uplink.A low-complexity power control algorithm was proposed to maximize the full-duplex D2D links throughput while guaranteeing the quality of service(QoS) requirements for full-duplex D2D users and CU.Numerical results show that the proposed algorithm can improve the full-duplex D2D links throughput.Moreover,the performance of full-duplex D2D communications depend on the QoS requirements of CU,the distance of D2D pair and self-interference cancelation amounts.【期刊名称】《电信科学》【年(卷),期】2017(033)003【总页数】7页(P1-7)【关键词】设备直通通信;全双工;功率控制【作者】赵季红;何强;曲桦;栾智荣【作者单位】西安邮电大学通信与信息工程学院,陕西西安710061;西安交通大学电子与信息工程学院,陕西西安710049;西安邮电大学通信与信息工程学院,陕西西安710061;西安交通大学电子与信息工程学院,陕西西安710049;西安交通大学电子与信息工程学院,陕西西安710049【正文语种】中文【中图分类】TN929随着局域应用和智能终端的不断涌现，近距离移动数据业务迅速增多，两种作为未来5G无线网络[1,2]中潜在的关键技术，即基于近距离传输的D2D（device-to-device，设备直通）通信技术[3]和无线全双工（full-duplex，FD）传输技术[4,5]被广泛讨论和研究。