Pub Date : 2003-09-21DOI: 10.1109/APCC.2003.1274387
M. A. Saeed, B. M. Ali, M. H. Habaebi
This paper discusses the transmission of the orthogonal frequency division multiplexing (OFDM) signal through the multipath fading indoor channel and its capability to combat the intersymbol interference (ISI) as well as its effective implementation with the discrete Fourier transform is described. The channel model used was based on Saleh-Valenzuela model with lognormal fading distribution of gain amplitudes. Simulation modules were developed and the effect of the multipath on the OFDM system performance with BPSK, QPSK, 16PSK, 64PSK, 16QAM, 64QAM, and 128QAM modulations was evaluated in terms of the bit error rate (BER) as a function of the energy per bit-to-noise ratio (EBNR). The influence of the number of carriers as well as the guard interval duration on the performance was also investigated. Simulations showed that the EBNR required to achieve a certain BER is significantly increased by 8-10 dB for dense multipath fading channels over that required in AWGN channels. These performance measures are useful for the design and assessment of high speed indoor wireless communication systems.
{"title":"Performance evaluation of OFDM schemes over multipath fading channels","authors":"M. A. Saeed, B. M. Ali, M. H. Habaebi","doi":"10.1109/APCC.2003.1274387","DOIUrl":"https://doi.org/10.1109/APCC.2003.1274387","url":null,"abstract":"This paper discusses the transmission of the orthogonal frequency division multiplexing (OFDM) signal through the multipath fading indoor channel and its capability to combat the intersymbol interference (ISI) as well as its effective implementation with the discrete Fourier transform is described. The channel model used was based on Saleh-Valenzuela model with lognormal fading distribution of gain amplitudes. Simulation modules were developed and the effect of the multipath on the OFDM system performance with BPSK, QPSK, 16PSK, 64PSK, 16QAM, 64QAM, and 128QAM modulations was evaluated in terms of the bit error rate (BER) as a function of the energy per bit-to-noise ratio (EBNR). The influence of the number of carriers as well as the guard interval duration on the performance was also investigated. Simulations showed that the EBNR required to achieve a certain BER is significantly increased by 8-10 dB for dense multipath fading channels over that required in AWGN channels. These performance measures are useful for the design and assessment of high speed indoor wireless communication systems.","PeriodicalId":277507,"journal":{"name":"9th Asia-Pacific Conference on Communications (IEEE Cat. No.03EX732)","volume":"76 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-09-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129102471","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2003-09-21DOI: 10.1109/APCC.2003.1274456
R. Kuroda, M. Katsuki, A. Otaka, N. Miki
QoS (Quality-of-Services) technologies are very important for services like video streaming and real-time communications because they require continuous fixed bandwidth. In this paper we propose a new "feedback and distribution method" that provides per-flow based QoS for large-scale networks. We enhanced the method's ability to control jitter when distributing streaming data to a large number of users. TCP traffic is burst-type traffic and is used for Web navigation and E-mail, and UDP traffic is constant streaming traffic used for real-time applications. A new method to provide ene-to-end bandwidth control for all users in a large-scale network where streaming services are expected to be very important commercially. The proposed method enabled streaming stable distribution because streaming packets that were susceptible to jitter were protected from burst-data by dividing the input for each different type of flow. Traffic bursts absorbed by setting the traffic measurement interval to an optimal length. Our simulation results showed that streaming data with high bit rates hardly affected TCP data. Therefore, the proposed method stably distributed streaming data, and furthermore, it satisfactorily managed a mixture of various types of data.
{"title":"Providing flow-based quality-of-service control in a large-scale network","authors":"R. Kuroda, M. Katsuki, A. Otaka, N. Miki","doi":"10.1109/APCC.2003.1274456","DOIUrl":"https://doi.org/10.1109/APCC.2003.1274456","url":null,"abstract":"QoS (Quality-of-Services) technologies are very important for services like video streaming and real-time communications because they require continuous fixed bandwidth. In this paper we propose a new \"feedback and distribution method\" that provides per-flow based QoS for large-scale networks. We enhanced the method's ability to control jitter when distributing streaming data to a large number of users. TCP traffic is burst-type traffic and is used for Web navigation and E-mail, and UDP traffic is constant streaming traffic used for real-time applications. A new method to provide ene-to-end bandwidth control for all users in a large-scale network where streaming services are expected to be very important commercially. The proposed method enabled streaming stable distribution because streaming packets that were susceptible to jitter were protected from burst-data by dividing the input for each different type of flow. Traffic bursts absorbed by setting the traffic measurement interval to an optimal length. Our simulation results showed that streaming data with high bit rates hardly affected TCP data. Therefore, the proposed method stably distributed streaming data, and furthermore, it satisfactorily managed a mixture of various types of data.","PeriodicalId":277507,"journal":{"name":"9th Asia-Pacific Conference on Communications (IEEE Cat. No.03EX732)","volume":"10 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-09-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114787731","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2003-09-21DOI: 10.1109/APCC.2003.1274270
Mijeong Yang, J. Hahm, Young-Sun Kim, Sangha Kim
This paper proposes an architecture of intermediate system-to-intermediate system (IS-IS) for the ATM based multiprotocol label switching (MPLS) system. IS-IS is a link state routing protocol designed to provide routing in network layer protocols with datagram service. IS-IS has favored scalability in the aspect of minimizing storage and computing in level 1 routers. Therefore, it is important to support IS-IS for the MPLS system used in backbone networks. We propose the architecture of IS-IS routing protocol and extensions for traffic engineering in MPLS system. We also describe its implementation and test results.
{"title":"Design and implementation of the IS-IS routing protocol with traffic engineering","authors":"Mijeong Yang, J. Hahm, Young-Sun Kim, Sangha Kim","doi":"10.1109/APCC.2003.1274270","DOIUrl":"https://doi.org/10.1109/APCC.2003.1274270","url":null,"abstract":"This paper proposes an architecture of intermediate system-to-intermediate system (IS-IS) for the ATM based multiprotocol label switching (MPLS) system. IS-IS is a link state routing protocol designed to provide routing in network layer protocols with datagram service. IS-IS has favored scalability in the aspect of minimizing storage and computing in level 1 routers. Therefore, it is important to support IS-IS for the MPLS system used in backbone networks. We propose the architecture of IS-IS routing protocol and extensions for traffic engineering in MPLS system. We also describe its implementation and test results.","PeriodicalId":277507,"journal":{"name":"9th Asia-Pacific Conference on Communications (IEEE Cat. No.03EX732)","volume":"103 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-09-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127335397","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2003-09-21DOI: 10.1109/APCC.2003.1274422
Xiyan Ma, Yi Wu, Z. Niu, T. Saito
The IEEE 802.11 standard supports the coexistence of asynchronous and time-bounded traffic by use of the two modes of medium access control (MAC) protocol termed as DCF (distributed coordination function) and PCF (point coordination function) respectively. The former is a random access scheme originally designed for delay-insensitive data applications, and the latter is based on polling mechanism which is more suitable for real-time services such as voice and video etc. In this paper, we focus on the performance of the packetized voice transmission using PCF mode in IEEE 802.11 wireless LAN while the DCF mode is used to provide minimum bandwidth available for data transfers. The effect of fragmentation threshold, echo cancellation and burst errors on network performance is theoretically analyzed.
{"title":"Performance analysis of the packetized voice transmission with PCF in an IEEE 802.11 infrastructure wireless LAN","authors":"Xiyan Ma, Yi Wu, Z. Niu, T. Saito","doi":"10.1109/APCC.2003.1274422","DOIUrl":"https://doi.org/10.1109/APCC.2003.1274422","url":null,"abstract":"The IEEE 802.11 standard supports the coexistence of asynchronous and time-bounded traffic by use of the two modes of medium access control (MAC) protocol termed as DCF (distributed coordination function) and PCF (point coordination function) respectively. The former is a random access scheme originally designed for delay-insensitive data applications, and the latter is based on polling mechanism which is more suitable for real-time services such as voice and video etc. In this paper, we focus on the performance of the packetized voice transmission using PCF mode in IEEE 802.11 wireless LAN while the DCF mode is used to provide minimum bandwidth available for data transfers. The effect of fragmentation threshold, echo cancellation and burst errors on network performance is theoretically analyzed.","PeriodicalId":277507,"journal":{"name":"9th Asia-Pacific Conference on Communications (IEEE Cat. No.03EX732)","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-09-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125446730","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2003-09-21DOI: 10.1109/APCC.2003.1274358
K. Oteng-Amoako, S. Nooshabadi
The paper presents an optimisation strategy based on discrete adaptation of the code rate given the modulation scheme in direct spread code division multiple access (DS- CDMA) communication system. The adaptive algorithm maximizes spectral efficiency across a fading channel by transmitting an optimal amount of incremental bits on each transmit attempts. The justification presented is based on a type-II hybrid-ARQ employing turbo punctured codes with multilevel signalling.
{"title":"Discrete adaptation of type-II hybrid-ARQ in DS-CDMA","authors":"K. Oteng-Amoako, S. Nooshabadi","doi":"10.1109/APCC.2003.1274358","DOIUrl":"https://doi.org/10.1109/APCC.2003.1274358","url":null,"abstract":"The paper presents an optimisation strategy based on discrete adaptation of the code rate given the modulation scheme in direct spread code division multiple access (DS- CDMA) communication system. The adaptive algorithm maximizes spectral efficiency across a fading channel by transmitting an optimal amount of incremental bits on each transmit attempts. The justification presented is based on a type-II hybrid-ARQ employing turbo punctured codes with multilevel signalling.","PeriodicalId":277507,"journal":{"name":"9th Asia-Pacific Conference on Communications (IEEE Cat. No.03EX732)","volume":"133 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-09-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124513063","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2003-09-21DOI: 10.1109/APCC.2003.1274327
D. Guo, Chau-Yun Hsu
This paper presents a new systematic method that combines parity check and cyclic coding technology to reduce the peak to mean envelope power ratio (PMEPR) of multicarrier transmission system. According our study, this new systematic method is much efficient than detail search method to find a set of good code word. The first procedure of the proposed method is to use parity check technology to reduce the peak power by more than half and generate a set of M-ary code words. The second procedure is to create a polynomial generator matrix from Galois fields, and than using a linear code set, multiplied by the matrix, to generate an m-ary cyclic codes set. Finally, the m-ary set of cyclic code is mapped into an M-ary set of cyclic codes to produce final PEP distribution. The results of this mapping reveal that PMEPR is reduced. We simulate this new method by the specific example of an eight-carriers multicarrier transmission system.
{"title":"Systematic reducing the PAPR of OFDM by cyclic coding","authors":"D. Guo, Chau-Yun Hsu","doi":"10.1109/APCC.2003.1274327","DOIUrl":"https://doi.org/10.1109/APCC.2003.1274327","url":null,"abstract":"This paper presents a new systematic method that combines parity check and cyclic coding technology to reduce the peak to mean envelope power ratio (PMEPR) of multicarrier transmission system. According our study, this new systematic method is much efficient than detail search method to find a set of good code word. The first procedure of the proposed method is to use parity check technology to reduce the peak power by more than half and generate a set of M-ary code words. The second procedure is to create a polynomial generator matrix from Galois fields, and than using a linear code set, multiplied by the matrix, to generate an m-ary cyclic codes set. Finally, the m-ary set of cyclic code is mapped into an M-ary set of cyclic codes to produce final PEP distribution. The results of this mapping reveal that PMEPR is reduced. We simulate this new method by the specific example of an eight-carriers multicarrier transmission system.","PeriodicalId":277507,"journal":{"name":"9th Asia-Pacific Conference on Communications (IEEE Cat. No.03EX732)","volume":"16 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-09-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121863345","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2003-09-21DOI: 10.1109/APCC.2003.1274375
A. Gandhi, S.S. Dhekane
In this paper we discuss the various basic speech coding techniques such as waveform coding, parametric coding and the quantization schemes. We also review the enhanced waveform interpolative coding technique in detail. The EWI coding technique for low bit rates with several enhancements like analysis by synthesis (AbS) optimization of slowly evolving waveform (SEW), rapidly evolving waveform (REW) parametrization, REW quantization, etc. proves to be very efficient for mobile communications. Also discussed are enhanced post-filtering and a novel pitch search technique for speech enhancement. The subjective test results have indicated that the quality of the 2.8 Kb/s EWI exceeds that of the G.723.1 at 5.3 Kb/s. Based on the results, we conclude that speech coding low bit-rates especially the EWI coder has enormous vistas in future 4G mobile systems, Internet telephony, LEO systems, etc.
{"title":"Speech coding at very low bit-rates for mobile communication","authors":"A. Gandhi, S.S. Dhekane","doi":"10.1109/APCC.2003.1274375","DOIUrl":"https://doi.org/10.1109/APCC.2003.1274375","url":null,"abstract":"In this paper we discuss the various basic speech coding techniques such as waveform coding, parametric coding and the quantization schemes. We also review the enhanced waveform interpolative coding technique in detail. The EWI coding technique for low bit rates with several enhancements like analysis by synthesis (AbS) optimization of slowly evolving waveform (SEW), rapidly evolving waveform (REW) parametrization, REW quantization, etc. proves to be very efficient for mobile communications. Also discussed are enhanced post-filtering and a novel pitch search technique for speech enhancement. The subjective test results have indicated that the quality of the 2.8 Kb/s EWI exceeds that of the G.723.1 at 5.3 Kb/s. Based on the results, we conclude that speech coding low bit-rates especially the EWI coder has enormous vistas in future 4G mobile systems, Internet telephony, LEO systems, etc.","PeriodicalId":277507,"journal":{"name":"9th Asia-Pacific Conference on Communications (IEEE Cat. No.03EX732)","volume":"99 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-09-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123161929","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2003-09-21DOI: 10.1109/APCC.2003.1274322
A. Razak, Mohamad Izani Zainal Abidin, R. Komiya
This paper discusses the pitch variation in Malay and English emotional voice samples for six emotion states. LP analysis is carried out to calculate the emotion features in speech and the results are observed in terms of average pitch, pitch range and jitter. Comparison between male and female pitch is also done. Based on the pitch variation analysis, language factor does not affect the acoustic correlates of emotional speech.
{"title":"Emotion pitch variation analysis in Malay and English voice samples","authors":"A. Razak, Mohamad Izani Zainal Abidin, R. Komiya","doi":"10.1109/APCC.2003.1274322","DOIUrl":"https://doi.org/10.1109/APCC.2003.1274322","url":null,"abstract":"This paper discusses the pitch variation in Malay and English emotional voice samples for six emotion states. LP analysis is carried out to calculate the emotion features in speech and the results are observed in terms of average pitch, pitch range and jitter. Comparison between male and female pitch is also done. Based on the pitch variation analysis, language factor does not affect the acoustic correlates of emotional speech.","PeriodicalId":277507,"journal":{"name":"9th Asia-Pacific Conference on Communications (IEEE Cat. No.03EX732)","volume":"15 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-09-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131398722","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2003-09-21DOI: 10.1109/APCC.2003.1274309
A. Teoh, S. Samad, A. Hussain
Authentication and privacy are crucial to Internet security due to its increasingly popular as the channel for information exchange, and the storage of sensitive network. In general, public key cryptography is applicable for both purposes but designing a high security authentication and strong privacy protected system still remain an open problem because of the weak usage of the passwords. The alternative options like biometrics are well suite for authentication because it is unique to each individual and cannot be lost, stolen, or recreated. A verification model incorporated with an appropriate algorithm is presented to facilitate on-line speaker verification for the web based access control application. The system allows for both the enrolment session and the actual verification to be done via a website, using a PC at the client side equipped with a multimedia microphone. Verification is performed at the server side to permit or deny the request for access. Initial evaluation of the prototype system is discussed.
{"title":"An Internet based speech biometric verification system","authors":"A. Teoh, S. Samad, A. Hussain","doi":"10.1109/APCC.2003.1274309","DOIUrl":"https://doi.org/10.1109/APCC.2003.1274309","url":null,"abstract":"Authentication and privacy are crucial to Internet security due to its increasingly popular as the channel for information exchange, and the storage of sensitive network. In general, public key cryptography is applicable for both purposes but designing a high security authentication and strong privacy protected system still remain an open problem because of the weak usage of the passwords. The alternative options like biometrics are well suite for authentication because it is unique to each individual and cannot be lost, stolen, or recreated. A verification model incorporated with an appropriate algorithm is presented to facilitate on-line speaker verification for the web based access control application. The system allows for both the enrolment session and the actual verification to be done via a website, using a PC at the client side equipped with a multimedia microphone. Verification is performed at the server side to permit or deny the request for access. Initial evaluation of the prototype system is discussed.","PeriodicalId":277507,"journal":{"name":"9th Asia-Pacific Conference on Communications (IEEE Cat. No.03EX732)","volume":"14 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-09-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131673002","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2003-09-21DOI: 10.1109/APCC.2003.1274395
Liang Xiao, Anchun Wang, Shidong Zhou, Yan Yao
This article presents a practical scheduler algorithm-multiple carrier proportional fairness (MCPF), to dynamically allocate resource for the orthogonal frequency division multiplex (OFDM) system in both frequency and time domain. We analyze its performance on the Doppler fading multipath channel with cocentric system studio and find that the throughput of MCPF is more than 60% greater compared with the fixed scheduling algorithm when there are more than 2 users. This algorithm satisfies proportional fair rule and its throughput performance increases with the number of system user, reaching 100% in 8 users system.
{"title":"A dynamic resource scheduling algorithm for OFDM system","authors":"Liang Xiao, Anchun Wang, Shidong Zhou, Yan Yao","doi":"10.1109/APCC.2003.1274395","DOIUrl":"https://doi.org/10.1109/APCC.2003.1274395","url":null,"abstract":"This article presents a practical scheduler algorithm-multiple carrier proportional fairness (MCPF), to dynamically allocate resource for the orthogonal frequency division multiplex (OFDM) system in both frequency and time domain. We analyze its performance on the Doppler fading multipath channel with cocentric system studio and find that the throughput of MCPF is more than 60% greater compared with the fixed scheduling algorithm when there are more than 2 users. This algorithm satisfies proportional fair rule and its throughput performance increases with the number of system user, reaching 100% in 8 users system.","PeriodicalId":277507,"journal":{"name":"9th Asia-Pacific Conference on Communications (IEEE Cat. No.03EX732)","volume":"15 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-09-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115011712","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
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