Relevant bibliographies by topics / Voice traffic

Contents

  1. Journal articles
  2. Dissertations / Theses
  3. Books
  4. Book chapters
  5. Conference papers
  6. Reports

Academic literature on the topic 'Voice traffic'

Author: Grafiati
Published: 4 June 2021
Last updated: 6 September 2023

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Journal articles on the topic "Voice traffic"

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Nisar, Kashif. "Fourth Stage of Voice Priority Queue for VoIP over WLANs." International Journal of Interdisciplinary Telecommunications and Networking 4, no. 2 (April 2012): 48–63. http://dx.doi.org/10.4018/jitn.2012040104.

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Voice over Internet Protocol (VoIP) is growing rapidly during this decade. VoIP is seen as a short-term and long-term transmission for voice and audio traffic and is moving on Wireless Local Area Networks (WLANs) based on IEEE 802.11 standards. Currently, packet scheduling algorithms like Weighted Fair Queuing (WFQ), was mainly designed to provide the bandwidth reservation. The Strict Priority (SP) is low-cost to maintain the delay sensitive voice traffic. Also, a number of research scheduling solutions have been proposed like General processor sharing (GPS), Deficit Round Robin (DRR), Contention-Aware Temporally fair Scheduling (CATS). Unfortunately, the current scheduling won’t be able to handle the VoIP packets properly and they have drawbacks over real-time applications. The objective of this research is to propose a Fourth Stage of Voice Priority Queue (VPQ) packet scheduling and algorithm to ensure more throughput, fairness and efficient packet scheduling for VoIP performance of queues and traffics. A new scheduler flexible which is capable of satisfying the VoIP traffic flows. Experimental topologies on NS-2 network simulator were analyzed for voice traffic.
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Noll, A. Michael. "Technical opinion: does data traffic exceed voice traffic?" Communications of the ACM 42, no. 6 (June 1999): 121–24. http://dx.doi.org/10.1145/303849.303871.

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Nisar, Kashif, and Suhaidi Hassan. "Mobility of Mobile Station in Voice Priority Queue Scheduler for VoIPWLANs." Journal of Communications and Computer Engineering 2, no. 3 (December 26, 2011): 28. http://dx.doi.org/10.20454/jcce.2012.249.

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Voice over Internet Protocol (VoIP) is growing rapidly during this decade. VoIP is seen as a short-term and long-trem transmission for voice and audio traffic. Meanwhile, VoIP is moving on Wireless Local Area Networks (WLANs) based on IEEE 802.11 standards. Currently, many packet scheduling algorithms like Weighted Fair Queuing (WFQ), was mainly designed to provide the bandwidth reservation. The Strict Priority (SP) is low-cost to maintain the delay sensitive voice traffic. Also, a number of research scheduling solutions have been proposed like General processor sharing (GPS), Deficit Round Robin (DRR), Contention-Aware Temporally fair Scheduling (CATS). Unfortunately, the current scheduling will not be able to handle the VoIP packets with the proper manner and they have some drawbacks over real-time applications. The objective of this research is to propose a new Voice Priority Queue (VPQ) packet scheduling and algorithm to ensure more throughput, fairness and efficient packet scheduling for VoIP performance of queues and traffics. A new scheduler flexible which is capable of satisfying the VoIP traffic flows. Experimental topologies on NS-2 network simulator were analyzed for voice traffic. Preliminary results show that this can achieve maximum and more accurate VoIP quality throughput and fairness index in mobility of mobile station. We verified and validated VPQ an extensive experimental simulation study under various traffic flows over WLANs.
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Zhuang Qiao, Shunliang Zhang, Liuqun Zhai, and Xiaohui Zhang. "Encrypted 5G over-the-top voice traffic classification using deep learning." ITU Journal on Future and Evolving Technologies 3, no. 3 (December 9, 2022): 779–92. http://dx.doi.org/10.52953/eyif3681.

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With the commercialization of fifth-generation (5G), the rapid popularity of mobile Over-The-Top (OTT) voice applications brings huge impacts on the traditional telecommunication voice call services. Tunnel encryption technology such as Virtual Private Networks (VPNs) allow OTT users to escape the supervision of network operators easily, which may cause potential security risks to cyberspace. To monitor harmful OTT applications in the context of 5G, it is critical to identify encrypted OTT voice traffic. However, there is no comprehensive study on typical OTT voice traffic identification. This paper mainly focuses on analyzing OTT voice traffic in the 5G network specifically. We propose employing Long Short-Term Memory (LSTM) and Convolutional Neural Networks (CNNs) to identify encrypted 5G OTT voice traffic, study the identification performance of used deep learning methods in three different scenarios. To verify the performance of the proposed approach, we collect 28 types of typical OTT and non-OTT voice traffic from the experimental 5G network. Experimental results prove the effectiveness and robustness of the proposed approach in encrypted 5G OTT voice traffic identification.
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Jadhav, Atharva. "Traffic Sign Detection and Voice Alert." International Journal for Research in Applied Science and Engineering Technology 11, no. 5 (May 31, 2023): 4283–88. http://dx.doi.org/10.22214/ijraset.2023.52651.

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Abstract: The Traffic Sign Detection and Voice Alert is an intelligent system designed to improve road safety by detecting traffic signs andproviding voice alerts to the driver. The project is based on computer vision and machine learning techniques, and it uses a camera to capture real-time images of the road and traffic signs. The system analyzes the images to detect traffic signs, such as speed limit signs, stop signs, and no entry signs, using deep learning algorithms. Once a traffic sign is detected, the system provides a voice alert to the driver, informing them of the sign's message. The system has several advantages over traditional traffic sign detection systems. Firstly, it is more accurate and reliable due to the use of deep learning algorithms. Secondly, it provides real-time alerts to the driver, which helps them to stay focused on the road and avoid accidents. Finally, it is cost-effective and can be easily installed in any vehicle.
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Prasad, V. V. S. Maruthy, and Shuba Swaminathan. "Voice Governed Traffic Light Control System." IETE Journal of Education 36, no. 2-3 (April 1995): 115–20. http://dx.doi.org/10.1080/09747338.1995.11415622.

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Mathieson, Lesley. "Voice disorders following road traffic accidents." Journal of Laryngology & Otology 111, no. 10 (October 1997): 903–6. http://dx.doi.org/10.1017/s0022215100138940.

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Greenstein, Larry. "Transporting voice traffic over packet networks." International Journal of Network Management 8, no. 4 (July 1998): 227–34. http://dx.doi.org/10.1002/(sici)1099-1190(199807/08)8:4<227::aid-nem290>3.0.co;2-p.

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Chang, Cheng-Shang, Xiuli Chao, and Michael Pinedo. "Integration of Discrete-Time Correlated Markov Processes in a TDM System." Probability in the Engineering and Informational Sciences 4, no. 1 (January 1990): 29–56. http://dx.doi.org/10.1017/s0269964800001443.

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In this paper, we consider a discrete-time queueing model for a Time Division Multiplexing (TDM) system with integration of voice and data (a model introduced by Li and Mark [16]). The voice traffic is a superposition of N Markov chains, which alternate between two states: the talkspurt state and the silence state. The data traffic is Poisson and independent of the voice sources. We show that the average queue size is increasing in certain correlation coefficients of the voice sources, increasing convex in the proportion of time the voice sources are in talkspurts, increasing convex in the number of voice sources, and increasing convex in the data traffic intensity. However, it is decreasing convex in the number of channels. These structural results yield various bounds. To take video traffic into account as well, we adapt a model of Maglaris et al. [18]. In their model, video traffic is generated by a continuous-state autoregressive Markov process that matches the average rate and the autocovariance of the output of a video coder. We show that if we replace their autoregressive model by a two-state Markov chain model with the same rate and correlation coefficient, we obtain an upper bound for the queue size. This replacement enables us to treat the video traffic as a voice source and use the techniques developed for dealing with voice/data integration to obtain bounds and estimates.
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Hong, Zhi Guo, Yong Bin Wang, and Min Yong Shi. "SPN-Based Performance Evaluation for ON-OFF Voice Traffic System." Advanced Materials Research 225-226 (April 2011): 1020–23. http://dx.doi.org/10.4028/www.scientific.net/amr.225-226.1020.

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By taking voice traffic system as study object, mathematical modeling methodology was used to investigate different parameters’ variations on system’s performance. Firstly, on the basis of analyzing the characteristics of voice traffic, this paper constructed a Stochastic Petri Net (SPN) model for ON-OFF voice traffic system. Furthermore, average time delay of the system was analyzed and model-based simulation was conducted with Stochastic Petri Net Package (SPNP) 6.0. For different parameters of ON to OFF rate and user’s voice connections, variation trends of number of user’s connections and ON to OFF rate are derived thereby. The methodology of modeling and simulation in this paper can be further used to analyze the performance of multimedia hybrid traffic system.
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Dissertations / Theses on the topic "Voice traffic"

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Radhakrishna, Deekonda, and Jannu Keerthipramukh. "OPNET simulation of voice over MPLS With Considering Traffic Engineering." Thesis, Blekinge Tekniska Högskola, Sektionen för datavetenskap och kommunikation, 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-3434.

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Multiprotocol Label Switching (MPLS) is an emerging technology which ensures the reliable delivery of the Internet services with high transmission speed and lower delays. The key feature of MPLS is its Traffic Engineering (TE), which is used for effectively managing the networks for efficient utilization of network resources. Due to lower network delay, efficient forwarding mechanism, scalability and predictable performance of the services provided by MPLS technology makes it more suitable for implementing real-time applications such as voice and video. In this thesis performance of Voice over Internet Protocol (VoIP) application is compared between MPLS network and conventional Internet Protocol (IP) network. OPNET modeler 14.5 is used to simulate the both networks and the comparison is made based on some performance metrics such as voice jitter, voice packet end-to-end delay, voice delay variation, voice packet sent and received. The simulation results are analyzed and it shows that MPLS based solution provides better performance in implementing the VoIP application. In this thesis, by using voice packet end-to-end delay performance metric an approach is made to estimate the minimum number of VoIP calls that can be maintained, in MPLS and conventional IP networks with acceptable quality. This approach can help the network operators or designers to determine the number of VoIP calls that can be maintained for a given network by imitating the real network on the OPNET simulator.
0046737675303
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Mennicke, Martin. "Simulation and interpretation for a voice-activated traffic information system." Honors in the Major Thesis, University of Central Florida, 2003. http://digital.library.ucf.edu/cdm/ref/collection/ETH/id/328.

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This item is only available in print in the UCF Libraries. If this is your Honors Thesis, you can help us make it available online for use by researchers around the world by following the instructions on the distribution consent form at http://library.ucf.edu/Systems/DigitalInitiatives/DigitalCollections/InternetDistributionConsentAgreementForm.pdf You may also contact the project coordinator, Kerri Bottorff, at kerri.bottorff@ucf.edu for more information.
Bachelors
Engineering
Computer Engineering
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Kamat, Narasinha. "A delay-efficient rerouting scheme for voice over ip traffic." [Gainesville, Fla.] : University of Florida, 2002. http://purl.fcla.edu/fcla/etd/UFE0000548.

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Al-Najjar, Camelia. "An approach for improving performance of aggregate voice-over-IP traffic." Texas A&M University, 2005. http://hdl.handle.net/1969.1/4216.

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The emerging popularity and interest in Voice-over-IP (VoIP) has been accompanied by customer concerns about voice quality over these networks. The lack of an appropriate real-time capable infrastructure in packet networks along with the threats of denial-of service (DoS) attacks can deteriorate the service that these voice calls receive. And these conditions contribute to the decline in call quality in VoIP applications; therefore, error-correcting/concealing techniques remain the only alternative to provide a reasonable protection for VoIP calls against packet losses. Traditionally, each voice call employs its own end-to-end forward-error-correction (FEC) mechanisms. In this paper, we show that when VoIP calls are aggregated over a provider's link, with a suitable linear-time encoding for the aggregated voice traffic, considerable quality improvement can be achieved with little redundancy. We show that it is possible to achieve rates closer to channel capacity as more calls are combined with very small output loss rates even in the presence of significant packet loss rates in the network. The advantages of the proposed scheme far exceed similar or other coding techniques applied to individual voice calls.
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Kharel, Jeevan, and Deepak Adhikari. "Performance Evaluation of Voice Traffic over MPLS Network with TE and QoS Implementation." Thesis, Blekinge Tekniska Högskola, Sektionen för datavetenskap och kommunikation, 2011. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-4757.

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Multiprotocol Label Switching (MPLS) is a new paradigm in routing architectures which has changed the way Internet Protocol (IP) packet is transferred in a Network. MPLS ensures the reliability of the communication minimizing the delays and enhancing the speed of packet transfer. One important feature of MPLS is its capability of providing Traffic Engineering (TE) which plays a vital role for minimizing the congestion by efficient load, balancing and management of the network resources. The performance evaluation is done considering the network parameters latency, jitter, packet end to end delay, and packet delay variation. Integration of QoS with the MPLS-TE network may enhance the performance of the network. Various scheduling algorithms can be used for implementing QoS on a network, which may vary the performance of the network. In our study, QoS is implemented on top of the MPLS-TE network using Differentiated Service (DiffServ) architecture. Different basic scheduling algorithms are used for the implementation of QoS and to check their impact on the network and to identify the suitable one among them. Performance evaluation is done considering the network parameters latency, jitter, packet end-to-end delay, and Packet Delay Variation. The simulation was done using OPNET modeler 16.0 and the results were analyzed. The simulation result shows that using TE along with QoS in MPLS network decreases the latency, jitter, packet delay variation and end to end packet delay compared to using TE alone for voice traffic.
+46738732963
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Nankani, Ajeet. "Horizontal Handoffs within WLANs : A detailed analysis and measurement concerning voice like traffic." Thesis, KTH, Mikroelektronik och Informationsteknik, IMIT, 2005. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-92271.

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IEEE 802.11 based Wireless Local Area Networks (WLANs) in addition to being used as access networks for providing traditional data services, are now also being used as access networks for providing realtime services such as VoIP and multimedia streaming. These realtime services are sensitive to latency, hence requiring seamless or low delay service from the lower layers throughout an ongoing session. The IEEE 802.11 standard does not define any technique or algorithm to provide seamless connectivity during the process of handoff, hence it does not require 802.11 based WLANs to provide the same. Thus, it is typical that there is a latency of 500 milliseconds to 1000 milliseconds during the handoff, before the mobile station can connect and receive data from the new access point (AP). However, many realtime services can not tolerate this much latency. The problem of handoff latency is further aggravated when WLANs are secured using IEEE 802.11i standard and when Authentication, Authorization & Accounting (AAA) services are involved in controlling network access to 802.11 based WLANs. This thesis will address the entire handoff process and examine the latency -- especially regarding AAA services. Different techniques and suggestions will be presented and analyzed closely at different layers and based on the results, an appropriate/efficient algorithm is suggested which will reduce this handoff latency, such that that seamless handoff can be achieved and realtime services can be provided over 802.11i enabled IEEE 802.11 WLANs.
Wireless Local Area Network (WLAN), baserat på IEEE 802.11 har traditionellt nyttjats som som accessnät för vanliga datatjänster. Ett allt vanligare användningsområde har blivit att nyttja samma nät för realtidstjänster som Voice over IP (VoIP) och mutimedia. Realtidstjänster är känsliga för fördröjningar. Fördröjningar som bland annat kan erhållas från de lägre nivåerna i OSI-stacken. IEEE 802.11-standarden definierar ingen teknik eller algoritm för att säkerställa avbrottsfri/fördröjningsfri transmission av data vid handoff och följdaktligen så kan man idag inte luta sig mot denna standard för att erhålla denna funktionalitet. Med nyttjande av befintlig IEEE 802.11 standarder erhålls fördröjningar på mellan 0,5 till 1 sekunder. Detta är naturligtvis inte acceptablet för många realtid och realtidsliknande tjänster. Problemet vid handoff accentueras ytterliggare om kravs ställs på AAA-tjänster för att säkerställa säkerheten i ett IEEE 802.11-baserat WLAN. Denna uppsats adresserar hela handoffprocessen med tillhörande fördröjningar – speciellt med hänsyn till AAA-tjänsterna. Olika tekninker och förslag presenteras och analyseras på olika nivåer. Baserat på erhållna resultat föreslås en algoritm för att reducera tidsåtgång vid handoff, så att realtidsliknande tjänster erhålls, utan störande fördröjningar, vid nyttjande av 802.11i.
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Shaikh, A. D. "Modelling data and voice traffic over IP networks using continuous-time Markov models." Thesis, Aston University, 2009. http://publications.aston.ac.uk/15385/.

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Common approaches to IP-traffic modelling have featured the use of stochastic models, based on the Markov property, which can be classified into black box and white box models based on the approach used for modelling traffic. White box models, are simple to understand, transparent and have a physical meaning attributed to each of the associated parameters. To exploit this key advantage, this thesis explores the use of simple classic continuous-time Markov models based on a white box approach, to model, not only the network traffic statistics but also the source behaviour with respect to the network and application. The thesis is divided into two parts: The first part focuses on the use of simple Markov and Semi-Markov traffic models, starting from the simplest two-state model moving upwards to n-state models with Poisson and non-Poisson statistics. The thesis then introduces the convenient to use, mathematically derived, Gaussian Markov models which are used to model the measured network IP traffic statistics. As one of the most significant contributions, the thesis establishes the significance of the second-order density statistics as it reveals that, in contrast to first-order density, they carry much more unique information on traffic sources and behaviour. The thesis then exploits the use of Gaussian Markov models to model these unique features and finally shows how the use of simple classic Markov models coupled with use of second-order density statistics provides an excellent tool for capturing maximum traffic detail, which in itself is the essence of good traffic modelling. The second part of the thesis, studies the ON-OFF characteristics of VoIP traffic with reference to accurate measurements of the ON and OFF periods, made from a large multi-lingual database of over 100 hours worth of VoIP call recordings. The impact of the language, prosodic structure and speech rate of the speaker on the statistics of the ON-OFF periods is analysed and relevant conclusions are presented. Finally, an ON-OFF VoIP source model with log-normal transitions is contributed as an ideal candidate to model VoIP traffic and the results of this model are compared with those of previously published work.
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Zhang, Shu Carleton University Dissertation Engineering Systems and computer. "Congestion control in frame relay networks with variable bit rate compressed voice and data traffic." Ottawa, 1993.

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Jamalipour, Abbas, Masaaki Katayama, Takaya Yamazato, and Akira Ogawa. "Performance of an Integrated Voice/Data System in Nonuniform Traffic Low Earth-Orbit Satellite Communication Systems." IEEE, 1995. http://hdl.handle.net/2237/7073.

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Om, Shree. "Voice and rural wireless mesh community networks: a framework to quantify scalability and manage end-user smartphone battery consumption." University of Western Cape, 2021. http://hdl.handle.net/11394/8174.

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Philosophiae Doctor - PhD
Community wireless mesh initiatives are a pioneering option to cheap ‘last-mile’ access to network services for rural low-income regions primarily located in Sub-Saharan Africa and Developing Asia. However, researchers have criticized wireless mesh networks for their poor scalability; and scalability quantification research has mostly consisted of modularization of per-node throughput capacity behaviour. A scalability quantification model to design wireless mesh networks to provide adequate quality of service is lacking. However, scalability quantification of community mesh networks alone is inadequate because rural users need affordable devices for access; and they need to know how best to use them. Low-cost low-end smartphones offer handset affordability solutions but require smart management of their small capacity battery. Related work supports the usage of Wi-Fi for communication because it is shown to consume less battery than 2G, 3G or Bluetooth. However, a model to compare Wi-Fi battery consumption amongst different low-end smartphones is missing, as is a comparison of different over-the-top communication applications.
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Books on the topic "Voice traffic"

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Voice teletraffic systems engineering. Norwood, MA: Artech House, 1988.

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United States. Office of Aviation Medicine., ed. An analysis of approach control/pilot voice communications. Washington, D.C: U.S. Dept. of Transportation, Federal Aviation Administration, Office of Aviation Medicine, 1996.

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FEDERAL AVIATION ADMINISTRATION. Project implementation plan, voice switching and control system. [Washington, D.C.?]: Dept. of Transportation, Federal Aviation Administration, 1994.

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Cardosi, Kim M. An analysis of tower (local) controller - pilot voice communications. Washington, D. C: U.S. Dept. of Transportation, Federal Aviation Administration, 1994.

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W, Britton Thomas, Hendrix Alfred M, Civil Aeromedical Institute, and United States. Office of Aviation Medicine., eds. Development of a coding form for approach control/pilot voice communications: Final report. Washington, D.C: Office of Aviation Medicine, U.S. Dept. of Transportation, 1995.

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United States. Office of Aviation Medicine. and Civil Aeromedical Institute, eds. An analysis of voice communication in a simulated approach control environment. Washington, D.C: U.S. Dept. of Transportation, Federal Aviation Administration, Office of Aviation Medicine, 1998.

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Office, General Accounting. Air traffic control: Voice communications system continues to encounter difficulties : report to congressional requesters. Washington, D.C: The Office, 1989.

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Veronika, Prinzo O., United States. Office of Aviation Medicine., Civil Aeromedical Institute, and University of New Hampshire, eds. Improving pilot/ATC voice communication in general aviation. Washington, D.C: U.S. Dept. of Transportation, Federal Aviation Administration, Office of Aviation Medicine, 1999.

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McEachern, Michael J. Voice-enabled applications over the Internet - voice XML interrogation of a web-enabled database for the provision of a traffic information service. Oxford: Oxford Brookes University, 2002.

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Chungguk tongbuk chiyŏk Han'guk kwallyŏn mayak pŏmjoe wa poisŭ p'ising pŏmjoe ŭi sit'ae mit taeŭng pangan e kwanhan yŏn'gu: The research and legal studies on the policy and the trends of drug and voice phishing crime related to Korean in northeast China. Sŏul T'ŭkpyŏlsi: Han'guk Hyŏngsa Chŏngch'aek Yon'guwŏn., 2014.

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Book chapters on the topic "Voice traffic"

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Verma, Pramode K., and Ling Wang. "Traffic Characterization." In Voice over IP Networks, 25–33. Berlin, Heidelberg: Springer Berlin Heidelberg, 2011. http://dx.doi.org/10.1007/978-3-642-14330-4_3.

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Aziz, Zagroz, and Robert Bestak. "Mobile Voice Traffic Load Characteristics." In Computer Networks, 193–207. Cham: Springer International Publishing, 2019. http://dx.doi.org/10.1007/978-3-030-21952-9_15.

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Sherine Mary, W., S. Muthukumar, K. S. Akilanayaki, N. Sathya Priya, and S. Sathish Kumar. "Implementation of Confidentiality for Voice Traffic." In International Conference on Intelligent Data Communication Technologies and Internet of Things (ICICI) 2018, 1187–93. Cham: Springer International Publishing, 2018. http://dx.doi.org/10.1007/978-3-030-03146-6_138.

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Aziz, Zagroz, and Robert Bestak. "Dependency Between the Distance and International Voice Traffic." In Ubiquitous Networking, 193–204. Cham: Springer International Publishing, 2020. http://dx.doi.org/10.1007/978-3-030-58008-7_16.

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Chong, Ilyoung, Chul-Woon Jang, and Hyun-Kook Kahng. "Voice Traffic Characterization Models in VoIP Transport Network." In Information Networking. Advances in Data Communications and Wireless Networks, 473–82. Berlin, Heidelberg: Springer Berlin Heidelberg, 2006. http://dx.doi.org/10.1007/11919568_47.

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Verma, Pramode K., and Ling Wang. "Impact of Bounded Delays on Resource Consumption in Packet-Switched Networks with M/M/1 Traffic." In Voice over IP Networks, 35–47. Berlin, Heidelberg: Springer Berlin Heidelberg, 2011. http://dx.doi.org/10.1007/978-3-642-14330-4_4.

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Kim, Jisoo, Daein Choi, Jungjin Park, Youn-Kwan Kim, I. Chong, and Hyun-Kook Kahng. "Performance of Voice Traffic over Mobile Ad Hoc Network." In Lecture Notes in Computer Science, 241–48. Berlin, Heidelberg: Springer Berlin Heidelberg, 2004. http://dx.doi.org/10.1007/978-3-540-25978-7_25.

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Gunawardena, Subodha, and Weihua Zhuang. "Service Response Time of Elastic Data Traffic." In Modeling and Analysis of Voice and Data in Cognitive Radio Networks, 41–69. Cham: Springer International Publishing, 2014. http://dx.doi.org/10.1007/978-3-319-04645-7_3.

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Gunawardena, Subodha, and Weihua Zhuang. "Service Response Time of Interactive Data Traffic." In Modeling and Analysis of Voice and Data in Cognitive Radio Networks, 71–85. Cham: Springer International Publishing, 2014. http://dx.doi.org/10.1007/978-3-319-04645-7_4.

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Tran, Hung Tuan, and Thomas Ziegler. "An Admission Control Scheme for Voice Traffic over IP Networks." In Lecture Notes in Computer Science, 353–64. Berlin, Heidelberg: Springer Berlin Heidelberg, 2003. http://dx.doi.org/10.1007/978-3-540-45076-4_36.

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Conference papers on the topic "Voice traffic"

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Pugaczewski, Jack T., Dave Sanderson, and David W. Hoffman. "ATM traffic-policing and traffic-shaping measurements." In Voice, Video, and Data Communications, edited by Wai Sum Lai and Hisashi Kobayashi. SPIE, 1997. http://dx.doi.org/10.1117/12.290414.

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Moh, W. Melody, and Neela Jorapur. "Advanced traffic-policing mechanisms for VBR traffic over ATM networks." In Voice, Video, and Data Communications, edited by Wai Sum Lai and Hisashi Kobayashi. SPIE, 1997. http://dx.doi.org/10.1117/12.290428.

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Allen, Mark E., and George M. Scheets. "Three-state traffic model and fitting technique for frame relay traffic." In Voice, Video, and Data Communications, edited by Wai Sum Lai and Hisashi Kobayashi. SPIE, 1997. http://dx.doi.org/10.1117/12.290433.

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Luoma, Marko, and Mika Ilvesmaeki. "Simplified management of ATM traffic." In Voice, Video, and Data Communications, edited by Wai Sum Lai and Hisashi Kobayashi. SPIE, 1997. http://dx.doi.org/10.1117/12.290431.

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Chatranon, Gwyn, and Sujata Banerjee. "Bandwidth allocation computations with measured traffic." In Voice, Video, and Data Communications, edited by Wai Sum Lai and Hisashi Kobayashi. SPIE, 1997. http://dx.doi.org/10.1117/12.290412.

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Iida, Katsuyoshi, Tetsuya Takine, Hideki Sunahara, and Yuji Oie. "Delay analysis for CBR traffic in static-priority scheduling: single-node and homogeneous CBR traffic case." In Voice, Video, and Data Communications, edited by Wai Sum Lai and Hisashi Kobayashi. SPIE, 1997. http://dx.doi.org/10.1117/12.290420.

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Park, Kihong, Gitae Kim, and Mark E. Crovella. "Effect of traffic self-similarity on network performance." In Voice, Video, and Data Communications, edited by Wai Sum Lai and Hisashi Kobayashi. SPIE, 1997. http://dx.doi.org/10.1117/12.290419.

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Yoo, Myungsik, Myoungki Jeong, and Chunming Qiao. "High-speed protocol for bursty traffic in optical networks." In Voice, Video, and Data Communications, edited by John M. Senior, Robert A. Cryan, and Chunming Qiao. SPIE, 1997. http://dx.doi.org/10.1117/12.290398.

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Mokhtar, Ahmed, and Murat Azizoglu. "Performance of packet-switched WDM broadcast networks with multicast traffic." In Voice, Video, and Data Communications, edited by John M. Senior, Robert A. Cryan, and Chunming Qiao. SPIE, 1997. http://dx.doi.org/10.1117/12.290382.

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Salinas-Hernandez, Ricardo R., Oscar R. Reyes, and Ramon M. Rodriguez-Dagnino. "Video-on-demand ATM system with MPEG VBR video traffic." In Voice, Video, and Data Communications, edited by Wai Sum Lai and Hisashi Kobayashi. SPIE, 1997. http://dx.doi.org/10.1117/12.290409.

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Reports on the topic "Voice traffic"

1

Floyd, S. IAB Concerns Regarding Congestion Control for Voice Traffic in the Internet. Edited by J. Kempf. RFC Editor, March 2004. http://dx.doi.org/10.17487/rfc3714.

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Thompson, Gardner W. An Entropy-based Approach to Detecting Anomalies in Voice over Internet Protocol (VoIP) Traffic. Fort Belvoir, VA: Defense Technical Information Center, March 2010. http://dx.doi.org/10.21236/ada532059.

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Asaturov, M. U., and G. K. Korendyasev. ENHANCEMENT OF VIBRATION STABILITY OF TERMINAL EQUIPMENT CASES OF VOICE COMMUNICATION CONTROL SYSTEMS (VCCS) FOR AIR TRAFFIC CONTROLLERS. DOI СODE, 2021. http://dx.doi.org/10.18411/vntr2021-163-1.

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