Dissertations / Theses on the topic 'QoS, VoIP'
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1
Islam, Mohammad Shahidul, and Syed Nasir Mehdi. "How Different QoS Mechanisms Affect VoIP QoS Metrics." Thesis, Högskolan i Halmstad, Sektionen för Informationsvetenskap, Data– och Elektroteknik (IDE), 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:hh:diva-15337.
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Voice over Internet Protocol (VoIP) has become a key technology of communication. Our work has been a practical implemenation of different scenarios to show that VoIP voice quality can be improved by adopting certain Quality of Service(QoS) measures such as classification, marking or queuing. It has been discussed that different QoS metrics like delay, packet loss and jitter could affect the voice quality of VoIP. To reduce the negative affects, one option is to implement certain QoS mechanisms with some set of configurations. For this purpose, Cisco IP phones have been configured in our topology with routers, switches, traffic generators, end stations and VoIP quality monitoring software called VQmanager. Tests have been divided into two sets. In one test a fixed bandwidth of 70 kbps is set while in the other test a random bandwidth is set with trafic generators unleashing packets of traffic. In both these tests further scenarios with configurations are worked out. They include no QoS, Auto Qos and Customized Qos mechanisms. Results have been indicative of top performance by the Customized QoS mechanism, in both sets of tests, followed by Auto QoS and no QoS mechanisms. It has been observed that a customized scenario could be a particular configuration to any organization’s needs and that will have the lowest delay, jitter and packet loss which are the main QoS metrics that impact the voice quality of VoIP. It can be fundamentally composed of classification of voice, data or web-traffic, marking and queuing depending upon the need of the organization. It is finally suggested to carry more tests in companies to get more data for analysis
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Liu, Mingkuan. "QoS Improvement Schemes for Real-Time Wireless VoIP." Diss., The University of Arizona, 2006. http://hdl.handle.net/10150/193858.
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There is a tremendous demand on real-time multimedia delivery over wireless Internet due to the dramatic increase in wireless communication and the growth of the Internet. However, real-time multimedia over wireless Internet poses many challenges. First of all, the inherent best-effort characteristic of packet-switched networks makes it difficult to provide guaranteed QoS for real-time multimedia delivery. Secondly, wireless channels have much higher packet-loss rate, bit-error rate, and channel instability compared to wired channels due to noise, path loss, multi-path fading and shadowing, which result in fluctuating communication channel statistics. Thirdly, the real-time communication demands strict time limitations on the network end-to-end delay and delay jitter.In this dissertation, an intelligent application architecture and several QoS improvement mechanisms are proposed to timely estimate the current wireless network statistics and dynamically take smart actions to improve the overall performance of a real-time wireless Internet telephony system. An online network traffic modeling method based on time series analysis was used to estimate the dynamic wireless network statistics such as end-to-end packet delay and delay jitters. Using this real-time updated information, the application's sender side can take some adaptive actions such as voice codec selection and forward error-correction schemes for packet-loss concealment to improve the QoS under current available network resources. Also, a novel adaptive playout jitter buffer adjustment algorithm is proposed. The proposed algorithm achieved 11%-15% performance improvement compared to traditional adaptive playout adjustment algorithms using the ITU-E model measurement metric.
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Simoni, Chiara. "Monitoraggio della QOS in interfacce wireless per voip." Bachelor's thesis, Alma Mater Studiorum - Università di Bologna, 2011. http://amslaurea.unibo.it/1917/.
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Saburova, S. A., Gmati Abdulbari, and О. I. Kadatskaja. "Methods Of Control Quality Of Services VoIP Over LTE." Thesis, ХНУРЕ, 2021. https://openarchive.nure.ua/handle/document/19003.
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Methods of control of providing users with VoIP over LTE network services are considered for the basic classes of LTE QoS in UMTS / 3GPP with the availability of the following traffic classes conversational, interactive and
streaming.Shown testing, calculation, analysis and evaluation of quality parameters for VoIP over LTE network services, the behavior of VoLTE MOS vs. packet loss Ppl and Pjitter, VoLTE MOS vs. effective packet loss VoLTE.
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Konečný, Zbyněk. "Mapování QoS požadavků na síťové prostředí." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2011. http://www.nusl.cz/ntk/nusl-218914.
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The issue of converged networks is to ensure the sufficient quality of services along the entire length of the communication transmission. This issue is closely connected to the real-time services, such as VoIP (Voice over Internet Protocol) and videoconferencing. These services require strict adherence to quality parameters, otherwise their function is not guaranteed. This problem particulary resolves subsystem IMS (IP Multimedia Subsystem), which concluded on the basis of user profiles can provide the required quality of service. Therefore the theoretical part deals with the description of the structure of the system and protocols designed to signal the network. Various mechanisms to support quality of services in access and backbone networks are also described. The following section explains the principle of provision of quality requirements of end-user networks. In the practical part is this theoretical knowledge used for designing and configuration of the network consisting of various technologies. The resulting model is then simulated in Opnet Modeler program, which is used for designing and testing of packet networks. Each simulation shows the effect of mapping quality requirements in the different access network on technologies, which are supported in the backbone. The outcome of this work is detailed network analysis and comparison of mechanisms for implementing quality of service. The conclusion summarises all simulation outcomes.
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Perera, Bandaralokuge Earl Shehan. "VoIP and best effort service enhancement on fixed WiMAX." Thesis, University of Canterbury. Electrical & Computer Engineering, 2008. http://hdl.handle.net/10092/1575.
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Fixed Broadband Wireless Access (BWA) for the last mile is a promising technology
which can offer high speed voice, video and data service and fill the technology gap
between Wireless LANs and wide area networks. This is seen as a challenging
competitor to conventional wired last mile access systems like DSL and cable, even in
areas where those technologies are already available. More importantly the
technology can provide a cost-effective broadband access solution in rural areas
beyond the reach of DSL or cable and in developing countries with little or no wired
last mile infrastructure. Earlier BWA systems were based on proprietary technologies
which made them costly and impossible to interoperate. The IEEE 802.16 set of
standards was developed to level the playing field. An industry group the WiMAX
Forum, was established to promote interoperability and compliance to this standard.
This thesis gives an overview of the IEEE 802.16 WirelessMAN OFDM standard
which is the basis for Fixed WiMAX. An in depth description of the medium access
control (MAC) layer is provided and functionality of its components explained.
We have concentrated our effort on enhancing the performance of Fixed
WiMAX for VoIP services, and best effort traffic which includes e-mail, web
browsing, peer-to-peer traffic etc. The MAC layer defines four native service classes
for differentiated QoS levels from the onset. The unsolicited grant service (UGS)
class is designed to support real-time data streams consisting of fixed-size data
packets issued at periodic intervals, such as T1/E1 and Voice over IP without silence
suppression, while the non-real-time polling service (nrtPS) and best effort (BE) are
meant for lower priority traffic. QoS and efficiency are at opposite ends of the scale in
most cases, which makes it important to identify the trade-off between these two
performance measures of a system. We have analyzed the effect the packetization
interval of a UGS based VoIP stream has on system performance. The UGS service
class has been modified so that the optimal packetization interval for VoIP can be
dynamically selected based on PHY OFDM characteristics. This involves cross layer
communication between the PHY, MAC and the Application Layer and selection of
packetization intervals which keep the flow within packet loss and latency bounds
while increasing efficiency. A low latency retransmission scheme and a new ARQ
feedback scheme for UGS have also been introduced. The goal being to guarantee
QoS while increasing system efficiency. BE traffic when serviced by contention based
access is variable in speed and latency, and low in efficiency. A detailed analysis of
the contention based access scheme is done using Markov chains. This leads to
optimization of system parameters to increase utilization and reduce overheads, while
taking into account TCP as the most common transport layer protocol. nrtPS is
considered as a replacement for contention based access. Several enhancements have
been proposed to increase efficiency and facilitate better connection management.
The effects of proposed changes are validated using analytical models in Matlab and
verified using simulations. A simulation model was specifically created for IEEE
802.16 WirelessMAN OFDM in the QualNet simulation package. In essence the aim
of this work was, to develop means to support a maximum number of users, with the
required level of service, using the limited wireless resource.
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Niemelä, Markus. "Estimating Internet-scale Quality of Service Parameters for VoIP." Thesis, Linköpings universitet, Programvara och system, 2016. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-127360.
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With the rising popularity of Voice over IP (VoIP) services, understanding the effects of a global network on Quality of Service is critical for the providers of VoIP applications. This thesis builds on a model that analyzes the round trip time, packet delay jitter, and packet loss between endpoints on an Autonomous System (AS) level, extending it by mapping AS pairs onto an Internet topology. This model is used to produce a mean opinion score estimate. The mapping is introduced to reduce the size of the problem in order to improve computation times and improve accuracy of estimates. The results of testing show that estimating mean opinion score from this model is not desirable. It also shows that the path mapping does not affect accuracy, but does improve computation times as the input data grows in volume.
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Rana, Bilal Zahid, and Shahid Ali. "OPNET Analysis of VoIP over MPLS VPN with IP QoS." Thesis, Blekinge Tekniska Högskola, Sektionen för datavetenskap och kommunikation, 2011. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-3404.
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There are many disadvantages (cost, lack of security, difficult to manage large networks, support to non-sensitive applications, delay, etc.) associated with traditional networking, IP network, ATM and Frame relay networking. To solve this, an MPLS-based VPN networking is introduced that can work with existing deployed backbones and allow organizations to interconnect the dispersed sites and remote workers through secure links by using public internet. In this thesis, we are trying to build a better understanding to MPLS VPN and we researched to analyze the behavior of OSPF and RIPv2 based MPLS-BGP VPN architectures by using intense VoIP traffic. Then it comes with an OPNET simulation process and scenarios for MPLS-BGP VPN. At last, the conclusion is made: OSPF based MPLS-BGP VPN architecture has lower VPN delay, background traffic Flow delay, LSP delay and point-to-point Queuing delay, and has better performance in VPN load and VPN throughput that can acquire customer satisfaction and confidence as compared to the RIPv2 based MPLS-BGP VPN architecture.Det finns många nackdelar (kostnader, bristande säkerhet, svåra att hantera stora nätverk, stöd till icke-känsliga tillämpningar, delay, etc.) i samband med traditionella nätverk, IP-nätverk, ATM och Frame Relay nätverk. För att lösa detta, är ett MPLS-baserat VPN nätverk införs som kan arbeta med befintliga sättas samman och låter organisationer för att förbinda de spridda platser och distansarbetare genom säkra länkar genom att använda publika Internet. I denna avhandling försöker vi bygga en bättre förståelse för MPLS VPN och vi forskat för att analysera beteendet hos OSPF och RIPv2 baserad MPLS-VPN BGP arkitekturer med hjälp av intensiv VoIP-trafik. Då kommer med en OPNET simulering process och scenarier för MPLS-BGP VPN. Äntligen är den slutsatsen: OSPF bygger MPLS-VPN BGP arkitektur har lägre VPN dröjsmål bakgrund trafikflödet dröjsmål, LSP dröjsmål och punkt-till-punkt Queuing dröjsmål, och har bättre prestanda i VPN-belastning och VPN som kan få kunden tillfredsställelse och förtroende jämfört med RIPv2 baserad MPLS-VPN BGP arkitektur.
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Paulsen, Stefan [Verfasser], and Bernd [Akademischer Betreuer] Wolfinger. "QoS/QoE-Modelle für den Dienst Voice over IP (VoIP) / Stefan Paulsen. Betreuer: Bernd Wolfinger." Hamburg : Staats- und Universitätsbibliothek Hamburg, 2016. http://d-nb.info/1106404505/34.
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Paulsen, Stefan Verfasser], and Bernd E. [Akademischer Betreuer] [Wolfinger. "QoS/QoE-Modelle für den Dienst Voice over IP (VoIP) / Stefan Paulsen. Betreuer: Bernd Wolfinger." Hamburg : Staats- und Universitätsbibliothek Hamburg, 2016. http://nbn-resolving.de/urn:nbn:de:gbv:18-79080.
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Holubovský, Petr. "Management výkonnosti a optimalizace VoIP technologie." Master's thesis, Česká zemědělská univerzita v Praze, 2016. http://www.nusl.cz/ntk/nusl-259801.
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The diploma thesis focuses on the VoIP technology optimization and performance management. The diploma thesis presents the theoretical basis of IP telephony and measurement of its quality. The thesis primarily deals with practical measurements of VoIP calls quality. Asterisk softswitch, various types of IP phones and simulated degradation of signal using Linux software router are used for measurements. Procedural diagram of VoIP technology real deployment is designed based on these measurements.
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Adnan, Muhammad. "QoS Analysis for Signaling in VoIP Client and Server Communication for Multicore." Thesis, KTH, Skolan för informations- och kommunikationsteknik (ICT), 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-93762.
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Due to the cost-effective solutions provided by Voice over Internet Protocol (VoIP) technology to enterprises and individuals, the growth has been significantly high in this area during the past and current decade. The growing demand has resulted in the escalating number of users who need secure, reliable and efficient communication systems. The deployment of multicore hardware has been solving the computational complexity problems. A multicore hardware/software model for VoIP is the key research area of modern telecommunications. One of the challenges is to design and implement a Quality of Service (QoS) benchmark module for multicore VoIP client and server environment. To achieve this we need a benchmarking module to quantitatively analyze QoS parameters namely delay and packet loss, and to further analyze these parameters with security overhead. In this project we have designed and implemented a prototype for a customized network traffic generator called SQgen, keeping in consideration the parallel nature of hardware and software in VoIP communication. The research focus area is to test the performance of signaling protocol for call set up process. Session Initiation Protocol (SIP) is widely deployed protocol for call establishment, maintenance and termination in VoIP, and we measure the performance of an open source implementation of SIP. Using SQgen, a series of stress tests are performed in different network scenarios to analyze the performance, and investigate the reasons for delays in different parts of the call setup process.
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Mushtaq, Muhammad Sajid, and Abdussalam Shahid. "QoS-Aware LTE Downlink Scheduler for VoIP in Relation With Power Saving." Thesis, Linköpings universitet, Institutionen för teknik och naturvetenskap, 2011. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-93123.
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The addition of multimedia services in cellular communication systems has created new challenges of resource allocation and power optimization. The requirement of efficient resource allocation is vital in downlink than uplink due to high traffic flows. These multimedia applications require more power therefore power optimization has gained a key role in future communication systems. This study investigates the performance of the downlink scheduling of Long Term Evolution (LTE) cellular communication network along efficient power utilization of User Equipment (UE). The goal is to develop a downlink scheduling technique that improves the QoS for multimedia services in relation to the use of power saving scheme i.e. Discontinuous Reception (DRX). The DRX effectively improves power consumption at the cost of QoSdegradation due to higher packet delays and packet losses. The traditional scheduling schemes were not designed to guarantee LTE QoS constraints in relation to energy. The proposed algorithm considers key QoS parameters during scheduling with fair resource allocation while minimizing packet delay and packet loss even in power saving environment. The performance of proposed scheduler with power saving technique is analyzed and its impact on QoS is evaluated in term of throughput, packet loss rate, fairness and packet delay. The proposed scheduler is compared with traditional scheduling algorithms such as Round Robin, Proportional Fair and Best CQI. The simulation results show that the proposed algorithm’s performance is better as compared to traditional scheduling algorithms in power saving and non-power saving environments.
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Cao, Jianguo, and j. cao@student rmit edu au. "An E-Model Implementation for VoIP QoS across a Hybrid UMTS Network." RMIT University. Electrical and Computer Engineering, 2009. http://adt.lib.rmit.edu.au/adt/public/adt-VIT20091028.134854.
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Voice over Internet Protocol (VoIP) provides a new telephony approach where the voice traffic passes over Internet Protocol shared traffic networks. VoIP is a significant application of the converged network principle. The research aim is to model VoIP over a hybrid Universal Mobile Telecommunications System (UMTS) network and to identify an improved approach to applying the ITU-T Recommendation G.107 (E-Model) to understand possible Quality of Service (QoS) outcomes for the hybrid UMTS network. This research included Modeling the hybrid UMTS network and carrying out simulations of different traffic types transmitted over the network. The traffic characteristics were analysed and compared with results from the literature. VoIP traffic was modelled over the hybrid UMTS network and the VoIP traffic was generated to represent different loads on the network from light to medium and heavy VoIP traffic. The VoIP over hybrid UMTS network traffic results were characterized and used in conjunction with the E-Model to identify VoIP QoS outcomes. The E-Model technique was implemented and results achieved were compared with results for other network types highlighted in the literature. The research identified an approach that permits accurate Modeling of VoIP QoS over a hybrid UMTS network. Accurate results should allow network design to facilitate new approaches to achieving an optimal network implementation for VoIP.
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Andersson, Martin. "Parametric Prediction Model for Perceived Voice Quality in Secure VoIP." Thesis, Linköpings universitet, Informationskodning, 2016. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-127402.
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More and more sensitive information is communicated digitally and with thatcomes the demand for security and privacy on the services being used. An accurateQoS metric for these services are of interest both for the customer and theservice provider. This thesis has investigated the impact of different parameterson the perceived voice quality for encrypted VoIP using a PESQ score as referencevalue. Based on this investigation a parametric prediction model has been developedwhich outputs a R-value, comparable to that of the widely used E-modelfrom ITU. This thesis can further be seen as a template for how to construct modelsof other equipments or codecs than those evaluated here since they effect theresult but are hard to parametrise. The results of the investigation are consistent with previous studies regarding theimpact of packet loss, the impact of jitter is shown to be significant over 40 ms.The results from three different packetizers are presented which illustrates theneed to take such aspects into consideration when constructing a model to predictvoice quality. The model derived from the investigation performs well withno mean error and a standard deviation of the error of a mere 1:45 R-value unitswhen validated in conditions to be expected in GSM networks. When validatedagainst an emulated 3G network the standard deviation is even lower.v
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Jánoš, Radan. "Řízení provozu na bezdrátových sítích." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2011. http://www.nusl.cz/ntk/nusl-218876.
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The Master´s thesis „Traffic management in wireless networks“ discusses how to ensure Quality of Service in these networks. A term „traffic management“ is connected mainly with certain restrictions and prioritization of some services and traffics in network. The thesis contains an overview of the most used wireless technologies and describes the approach of these technologies to ensure QoS. Theoretical part of the thesis follows with the definition of general principles of traffic management in IP networks and provides an overview of network parameters used in evaluating the quality of different types of communications services. Practical part is focused on the most widely used wireless technology of standard 802.11 and also on implementation of remote administration system on a small SOHO routers. These systems allow the use of queueing disciplines to manage QoS. Designed and implemented HTB discipline is tested on a real traffic network model.
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Ram, Abhishek. "Assessment of Voice Over IP as a solution for Voice over ADSL." Thesis, Virginia Tech, 2002. http://hdl.handle.net/10919/33135.
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Voice over DSL (VoDSL) is a technology that enables the transport of data and multiple voice calls over a single copper-pair. VoDSL employs packet voice technology instead of the traditional circuit switched voice. Voice over ATM (VoATM) and Voice over IP (VoIP) are the two main alternatives for carrying voice packets over DSL. ATM is currently the preferred technology, since it offers the advantage of ATMâ s built-in Quality of Service (QoS) mechanisms. IP, on the other hand, cannot provide QoS guarantees in its traditional form. IP QoS mechanisms have been evolved only in the recent years. VoIP has gained popularity in the core networks. If it could replace VoATM in the access networks, it would open the door for end-to-end IP telephony that would result in major cost savings. In this thesis, we propose a VoIP-based VoDSL architecture that provides QoS guarantees comparable to those offered by ATM in the DSL access network.
Our QoS architecture supports Premium and Regular service categories for voice traffic and the Best-Effort service category for data traffic. Voice and data packets are placed in separate output queues at the bottleneck link. The Weighted Fair Queuing algorithm in used to schedule voice and data packets for transmission over the bottleneck link. Fragmentation of large data packets reduces the waiting time for voice packets in the link. We also propose a new admission control mechanism called Admission Control by Implicit Signaling. This mechanism takes advantage of application layer signaling by mapping it to the IP header. The router can infer the resource requirements for the connection by looking at certain field in the IP header of the application layer signaling packets. This eliminates the need for an explicit signaling protocol.
We evaluate the performance of our QoS architecture by means of a simulation study. Our primary metrics are the end-to-end delay of voice packets across the access network and the bandwidth consumed by a voice call. Our results show that the end-to-end delays of voice packets in our VoIP architecture are comparable to that in the VoATM architecture. ACIS limits the number of voice calls admitted into the premium service class and provides guaranteed service to those calls under all loads. It also provides acceptable service to regular calls under light loads. We also show that PPP is a better choice than ATM as a Layer 2 protocol for our VoIP architecture. PPP offers the advantages of low bandwidth requirement and interleaving of voice packets in between fragments of large data packets during transmission over the bottleneck link. We conclude that our VoIP architecture would be suitable for future VoDSL deployments.Master of Science
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Bellalta, Boris. "Flow-level QoS guarantees in IEEE 802.11e EDCA-based WLANs." Doctoral thesis, Universitat Pompeu Fabra, 2007. http://hdl.handle.net/10803/7539.
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Les xarxes WLANs possibiliten un accés de banda ampla a Internet des d'un terminal mòbil, essent una possible solució alternativa a les
xarxes cel·lulars. Tanmateix, aquest tipus de tecnologia presenta certes
limitacions, com és la difícil coexistència entre fluxos de tràfic
rígids (VoIP) i fluxos de tràfic elàstic (TCP), degut al propi protocol
d'accés al medi. En aquesta tesi es proposa la utilització d'un nou
mecanisme de control d'admissió que utilitzant el nou estàndard de
qualitat de servei en xarxes WLAN (EDCA). La utilització del mecanisme
de control d'admissió millora notablement les prestacions que es poden
obtenir de la xarxa, solucionant les diferents limitacions de la
tecnologia. Per a l'avaluació i optimització del mecanisme de control
d'admissió s'ha desenvolupat un conjunt d'eines matemàtiques que
permeten capturar tant la dinàmica del protocol d'accés com el
comportament dels diferents fluxos de tràfic multimèdia que s'han
consideren (VoIP i tràfic elàstic, TCP).
WLANs provide a broadband access to Internet from a mobile terminal,
which can be a possible alternative solution to cellular networks.
However, this technology presents several limitations, as it is the
difficult coexistence between rigid traffic flows (VoIP) and elastic
traffic flows (TCP), due to the medium access protocol itself. In this
thesis a new admission control mechanism is proposed. It uses the set of
QoS mechanisms provided by the new EDCA standard. The use of the
proposed admission control mechanism improves the overall WLAN
performance, solving the different technology limitations. In order to
be able to evaluate and optimize the admission control mechanism,
several mathematical tools have been developed in order to capture the
dynamics of both, the access protocol and of the different multimedia
traffic flows that have been considered.
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Sfairopoulou, Anna. "A cross-layer mechanism for QoS improvements in VoIP over multi-rate WLAN networks." Doctoral thesis, Universitat Pompeu Fabra, 2008. http://hdl.handle.net/10803/7563.
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In IEEE 802.11 WLANs, Link Adaptation mechanisms, which choose the transmission rate of each node, provoke unexpected and random variations on the effective channel capacity. When these changes are towards lower bitrates, inelastic flows, such as VoIP, can suffer from sudden congestion, which results on higher packet delays and losses. In this thesis, a VoIP codec adaptation algorithm is proposed as a solution, based on a cross-layer feedback from RTCP packets and the MAC layer, which can adapt the codecs of active calls to adjust them to the multirate scenario. A combination of this algorithm with a call admission control mechanism is also studied. The results show an important improvement in terms of the QoS of the already active flows as also in the total hotspot's capacity. Additionally, by defining a new Grade of Service related parameter, the Q-Factor, which captures the trade-off between dropping and blocking ratio and perceived speech quality, the codec adaptation algorithm can be tuned to achieve maximum capacity without severely penalizing any of those variables, and hence satisfying both technical and user quality requirements. Finally, a new QoS-enabled AP, which implements these enhancements is designed.En las redes inalámbricas del estándar IEEE 802.11, los mecanismos de adaptación de enlace que eligen la tasa de transmisión de cada nodo, pueden provocar variaciones aleatorias e inesperadas en la capacidad efectiva del canal. Cuando estos cambios son hacia tasas de transmisión mas bajas, los flujos inelásticos, tales como los de VoIP, pueden de repente sufrir congestión, lo que se traduce en aumento de retrasos y pérdidas de paquetes. En esa tesis, se propone un algoritmo de adaptación de codificadores de voz como solución, basado en técnicas multinivel (cross-layer) que combinan el uso de información de diferentes capas, como los paquetes RTCP y la capa MAC, y que puede adaptar los codecs de las llamadas activas para ajustarlos al escenario "multi-rate". Adicionalmente, la combinación de este algoritmo con un mecanismo de control de admisión de llamadas (CAC) se ha estudiado. Los resultados muestran una importante mejora en términos de QoS de los flujos activos como también en la capacidad total del hotspot. Además, mediante la definición de un nuevo factor, el Q-Factor, que puede captar la compensación entre la tasa de corte y de bloqueo de llamadas y de la calidad percibida por esas, el algoritmo de adaptación de codecs se puede ajustar para lograr la máxima capacidad sin penalizar severamente ninguna de esas variables y así satisfacer los requisitos técnicos de calidad y los usuarios. Por último, un nuevo punto de acceso (AP) habilitado para ofrecer calidad de servicio, ha sido diseñado que lleva a cabo estas mejoras.
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Lewis, Rosemary. "Operational benefit of implementing VoIP in a tactical environment." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2003. http://library.nps.navy.mil/uhtbin/hyperion-image/03Jun%5FLewis.pdf.
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Thesis (M.S. in Information Systems and Operations)--Naval Postgraduate School, June 2003.Thesis advisor(s): Dan C. Boger, Rex Buddenberg. Includes bibliographical references (p. 61-62). Also available online.
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Monks, Eduardo Maronas. "Planejamento de capacidade em redes corporativas para implementação de serviços VoIP." reponame:Biblioteca Digital de Teses e Dissertações da UFRGS, 2006. http://hdl.handle.net/10183/13660.
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Este trabalho tem como objetivo o estudo da tecnologia VoIP (Voz sobre IP) e a sua aplicação em redes corporativas, enfocando o planejamento de capacidade da rede de dados para absorver serviços VoIP. Serão apresentados tópicos sobre a fundamentação teórica de VoIP (Voz sobre IP), os requisitos de arquitetura de rede e QoS (Qualidade de Serviço) exigidos pelo serviço. Mostra-se também como a metodologia para planejamento de capacidade usado em telefonia convencional pode ser adaptada aos serviços VoIP em uma rede corporativa. Foi aplicada a metodologia adaptada através de um estudo de caso em uma rede corporativa real.This work has as objective the study of capacity planning in corporate networks for the implementation of VoIP (Voice over IP) services. We will presents topics about the theorical background of VoIP, the requirements of architecture of network and QoS (Quality of Service) demanded by the service. It will also reveal how the methodology used for planning capacity in conventional telephony, could be adjusted to the VoIP services in a corporate network. The adjusted methodology was applied in a real corporate network.
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Novák, Přemek. "Optimalizace QoS a analýza závislostí komunikačních služeb na zpoždění." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2015. http://www.nusl.cz/ntk/nusl-220410.
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This thesis consists of theoretical and practical parts. The theoretical part contains an analysis of the problems of wireless networkssolutions individual standards and methodologies to ensure quality of service. In the practical part using the OPNET Modeler, a number of different simulation models and their evaluation. It is a model of individual standards, support mobility and interference effects on quality of service.
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Potfay, Attila. "Směrovací protokoly pro MANET sítě se zaměřením na QoS." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2013. http://www.nusl.cz/ntk/nusl-220206.
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The main task of this master’s thesis is to simulate the routing protocol AODV (Adhoc On-Demand Distance Vector Routing) to the Network Simulator ns-3 environment, and to realize a model of MANET (Mobile Ad-hoc Network) network with the support of Quality Of Service (QoS). Further implement the protocol AODV in real devices and the involvement of such nodes in the simulation process using the simulator ns-3. This work provides the theoretical basics: it deals with the primary characteristics of MANET, describes in detail the routing protocols MANET with support of QoS, provides information about the Network Simulator ns-3 simulation environment, describes in detail the existing implementation solutions of the protocol AODV in to real devices and provides information about the methods of involvement real devices to the simulation.
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Awan, Naser Saeed. "Characterization of SIP Signaling-Messages Over OpenSIPS Running On Multicore Server." Thesis, KTH, Skolan för informations- och kommunikationsteknik (ICT), 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-121530.
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Over the course of last decade, the demand for VoIP (Voice over Internet Protocol) applications has increased significantly among enterprises and individuals due to its low cost. This increasing demand resulted in a significant increase in users who require reliable VoIP communication systems. QoS (Quality of Service) is a major issue in VoIP implementation and is a method to impel the development of real-time multimedia services like VoIP and videoconferencing. However, there are certain challenges in achieving QoS for VoIP application, which need special attentions; like latency and packet loss. The VoIP servers which are functioning on single core software/hardware model have high latency and packet loss issues due to their limited processing bandwidth. A multicore software/hardware model is the solution to cope up with the increasing demands of VoIP and yet an active research area in telecommunication. Using a multicore software/hardware model for VoIP has several challenges, one of the challenges is to design and implement QoS Benchmarking module for VoIP client and server on multicore. In this thesis the focus is on latency and packet loss of SIP messages on OpenSIPS server. This is done by performing stress testing for QoS benchmarking, where delay and call drop rate is calculated for SIP (Session Initiation Protocol) signaling messages on parallel VoIP client server model. The model is built in C for multicore and is used as a simulation tool. SIP is widely deployed protocol for call establishment, maintenance and termination in VoIP.
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Nocito, Carlos Daniel. "A Network Conditions Estimator for Voice Over IP Objective Quality Assessment." Scholarly Repository, 2011. http://scholarlyrepository.miami.edu/oa_theses/292.
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Objective quality evaluation is a key element for the success of the emerging Voice over IP (VoIP) technologies. Although there are extensive economic incentives for the convergence of voice, data, and video networks, packet networks such as the Internet have inherent incompatibilities with the transport of real time services. Under this paradigm, network planners and administrators are interested in ongoing mechanisms to measure and ensure the quality of these real time services. Objective quality assessment algorithms can be broadly divided into a) intrusive (methods that require a reference signal), and b) non intrusive (methods that do not require a known reference signal). The latter group, typically requires knowledge of the network conditions (level of delay, jitter, packet loss, etc.), and that has been a very active area of research in the past decade. The state of the art methods for objective non-intrusive quality assessment provide high correlations with the subjective tests. Although good correlations have been achieved already for objective non-intrusive quality assessment, the current large voice transport networks are in a hybrid state, where the necessary network parameters cannot easily be observed from the packet traffic between nodes. This thesis proposes a new process, the Network Conditions Estimator (NCE), which can serve as bridge element to real-world hybrid networks. Two classifications systems, an artificial neural network and a C4.5 decision tree, were developed using speech from a database collected from experiments under controlled network conditions. The database was composed of a group of four female speakers and three male speakers, who conducted unscripted conversations without knowledge about the details of the experiment. Using mel frequency cepstral coefficients (MFCCs) as the feature-set, an accuracy of about 70% was achieved in detecting the presence of jitter or packet loss on the channel. This resulting classifier can be incorporated as an input to the E-Model, in order to properly estimate the QoS of a network in real time. Additionally, rather than just providing an estimation of subjective quality of service provided, the NCE provides an insight into the cause for low performance.
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Zelinka, Jiří. "Podpora kvality služeb v koncových aplikacích." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2008. http://www.nusl.cz/ntk/nusl-217440.
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The topic of this diploma thesis has been chosen for a discussion and implementation of quality of service on the application and link layer of the OSI model. At the beginning of this thesis the general parameters and basic technologies of the OSI model network layer has been explained. This part was chosen as a resource for explanation of the basic parameters in the quality of services. Basic part is followed by the introduction to the Ethernet technology, which became as a ground for the real model in this diploma thesis. As a part of this section has been written a block which contains a analysis of quality of services in the Ethernet, which means the implementation of IEEE 802.1Q/p. This analysis is followed by structured descrition of the controlling and functionality of quality of services parameters on the link layer with the system tool in the Windows XP, description of Win XP link layer drivers and its modification with system tool tcmon. The end of the theoretical part is represented by introduction to the implementation of quality of service in the wireless networks especially the 802.11e standard. At the beginning of the practical part is specified the description of the topology design which is dedicated to the quality of services implementation. This section is developed to well-founded analysis of applications which were used during topology creating. The last segment of this thesis is dedicated to evolving the practical informations which were obtained during the measurement.
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Iqbal, Arshad. "VoIP Server HW/SW Codesign for Multicore Computing." Thesis, KTH, Skolan för informations- och kommunikationsteknik (ICT), 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-94203.
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Modern technologies are growing and Voice over Internet Protocol (VoIP) technology is able to function in heterogeneous networks. VoIP gained wide popularity because it offers cheap calling rates compared to traditional telephone system and the number of VoIP subscribers has increased significantly in recent years. End users need reliable and acceptable call quality in real time communication with best Quality of Service (QoS). Server complexity is increasing to handle all client requests simultaneously and needs huge processing power. VoIP Servers will increase processing power but the engineering tradeoff needs to be considered e.g. increasing hardware will increase hardware complexity, energy consumption, network management, space requirement and overall system complexity. Modern System-on-Chip (SoC) uses multiple core technology to resolve the complexity of hardware computation. With enterprises needing to reduce overall costs while simultaneously improving call setup time, the amalgamation of VoIP with SoC can play a major role in the business market. The proposed VoIP Server model with multiple processing capabilities embedded in it is tailored for multicore hardware to achieve the required result. The model uses SystemC-2.2.0 and TLM-2.0 as a platform and consists of three main modules. TLM is built on top of SystemC in an overlay architectural fashion. SystemC provides a bridge between software and hardware co-design and increases HW & SW productivity, driven by fast concurrent programming in real time. The proposed multicore VoIP Server model implements a round robin algorithm to distribute transactions between cores and clients via Load Balancer. Primary focus of the multicore model is the processing of call setup time delays on a VoIP Server. Experiments were performed using OpenSIP Server to measure Session Initiation Protocol (SIP) messages and call setup time processing delays. Simulations were performed at the KTH Ferlin system and based on the theoretical measurements from the OpenSIP Server experiments. Results of the proposed multicore VoIP Server model shows improvement in the processing of call setup time delays.
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Carvalho, Leandro Silva Galvão de. "Gerenciamento adaptativo da qualidade da fala entre terminais VoIP." Universidade Federal do Amazonas, 2011. http://tede.ufam.edu.br/handle/tede/3144.
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Previous issue date: 2011-10-07Voice calls based on Voice over Internet Protocol (VoIP) technology are liable to several impairments from both application and network layer, such as codec compression, end-to-end delay, and packet loss. For years, this problem has been challenging researchers and practitioners, who have been designing and improving QoS control mechanisms for VoIP applications. Such mechanisms aim to make optimum use of network and terminal resources so as to minimize the effects of network impairments on voice quality. Among the several proposed QoS control mechanisms for VoIP, some of them seek to adapt the voice flow or other VoIP-related parameters in accordance with significant changes in the network, end users preferences, or service providers requirements. VoIP systems are particularly likely to require a dynamic adaptation solution for dealing with the complex trade-off between speech quality and impairments, because of the decentralized control nature of IP networks and the stochastic nature of data packet delivery. Although the existing adaptive solutions for QoS control of VoIP show some performance improvement and exhibit some sort of feedback, they do not provide explicit focus on the control loop. This document shows the current progress of our thesis, which addresses the adjustment of internal parameters of VoIP terminals (at application layer) that affect the voice flow, with the aim of improving speech quality in response to changes in network conditions. It is not in the scope of the thesis to propose adaptive solutions that focus exclusively on signaling, billing, security issues, or operate at the network layer. Therefore, this thesis addresses the problem of how adjust encoding parameters in response to variations in delay and packet loss, in order to optimize speech quality. The objective is to optimize user-perceptible attributes of speech, under the perspective of self-adaptive software systems. The emphasis is not to develop new audio codecs, but to build a control loop in the core of sender and receiver terminals to adapt voice flow settings according to network conditions. The main contributions of this thesis are the following: determination of user s perception during codec switching; parametrization of codec precedence for supporting codec switching decision; explicit design of a monitoring analysis planning execution control loop as the core of the adaptation process; and efficiency analysis of feedback message exchanging.
Chamadas de voz baseadas na tecnologia VoIP (Voice over Internet Protocol) estão suscetíveis a degradações diversas, provenientes tanto da camada de aplicação, como da camada de rede, tais como compressão do codec, atraso fim a fim e perda de pacotes. Durante anos, esse problema tem desafiado pesquisadores e profissionais, que têm concebido e melhorado mecanismos de controle de QoS para aplicações VoIP. Tais mecanismos visam otimizar a utilização dos recursos da rede e do terminal VoIP de modo a minimizar os efeitos deletérios da rede subjacente sobre a qualidade de voz. Entre as várias propostas de mecanismos de controle de QoS para VoIP, alguns deles procuram adaptar o fluxo de voz ou outros parâmetros VoIP de acordo com mudanças significativas na rede, preferências de usuário, ou requisitos dos provedores de serviços VoIP. Sistemas VoIP particularmente exigem soluções de adaptação dinâmica para lidar com a complexa relação de compromisso entre qualidade de voz e fatores de degradação, por causa da natureza descentralizada e estocástica das redes IP na entrega de pacotes de voz. Embora as soluções adaptativas existentes para controle de QoS em VoIP mostrem alguma melhora de desempenho e apresentem algum tipo de feedback, elas não fornecem foco explícito na ciclo de controle (control loop). Este documento mostra o progresso atual da nossa tese, que aborda o ajuste de parâmetros internos de terminais VoIP (camada de aplicação) que afetam o fluxo de voz, com o objetivo de melhorar a qualidade da fala em resposta a mudanças nas condições da rede. Não faz parte do escopo da tese abordar soluções adaptativas que se concentram exclusivamente em sinalização, bilhetagem, problemas de segurança, ou que operam no nível da camada de rede. Portanto, esta tese aborda o problema da concepção e avaliação de estratégias adaptativas que explorem as relações de compromisso entre qualidade da fala e os seguintes fatores de degradação: compressão do codec, atraso fim a fim e perda de pacotes. A finalidade é otimizar atributos da fala perceptíveis aos usuário, sob a perspectiva de sistemas de software autoadaptativo. A ênfase não reside em desenvolver novos codecs de áudio, mas sim em desenvolver um ciclo de controle como entidade central de um terminal VoIP, que possa adaptar as configurações do fluxo de voz de acordo com as condições da rede. As principais contribuições desta tese são as seguintes: determinação da percepção do usuário durante a comutação de codec; parametrização de precedência de codecs para suporte de decisão de comutação de codec; enfoque no ciclo de controle baseado nas atividades de monitoramento análise planejamento execução como núcleo do processo de adaptação; e análise de eficiência de troca de mensagens de feedback.
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Janczukowicz, Ewa Czeslawa. "QoS management for WebRTC : loose coupling strategies." Thesis, Ecole nationale supérieure Mines-Télécom Atlantique Bretagne Pays de la Loire, 2017. http://www.theses.fr/2017IMTA0010/document.
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Depuis plusieurs années, on observe une multiplication des services de communication en temps réel de type Over-The-Top (OTT). Ces solutions utilisent l¿Internet « best-effort » et s¿adaptent aux fluctuations du réseau. Néanmoins, il est discutable que l¿approche OTT soit suffisante pour fournir une qualité de service de communication acceptable quelles que soient les conditions réseaux. Dès lors, est-il possible d¿utiliser l¿assistance réseau pour améliorer la qualité de service des solutions OTT ?Pour traiter cette question, nous étudions tout d¿abord les solutions OTT, et particulièrement la technologie WebRTC. Nous identifions trois stratégies de couplage lâche qui permettent de tirer parti des mécanismes réseaux pour améliorer la qualité de service des solutions OTT.Nous vérifions la pertinence de ces stratégies dans le contexte de la gestion du trafic. On identifie deux approches de gestion du trafic adaptées à WebRTC : 1) qui assure des délais d¿attente courts quel que soit le trafic ou 2) qui isole le trafic sensible. On évalue ces solutions et leur impact sur WebRTC, pour les réseaux d¿accès filaire (uplink, ADSL et fibre). Les résultats obtenus montrent que les pratiques actuelles de gestion du trafic ne sont pas adaptées au trafic WebRTC. De plus, les solutions proposées assurent plus d¿équité entre le trafic WebRTC et TCP et elles permettent d¿éviter que le trafic WebRTC soit désavantagé et elles améliorent la qualité de communication.Enfin, ces solutions de la gestion du trafic sont positionnées dans le contexte des stratégies de couplage proposées. A partir de là, on fournit des recommandations pour améliorer la qualité WebRTC avec l¿assistance du NSPThe number of real-time Over-The-Top (OTT) communication services has increased in the recent years. OTT solutions use the best-effort Internet delivery and rely on mechanisms built into the endpoints to adapt to underlying network fluctuations. Nevertheless, it is questionable if this approach is enough to provide acceptable quality of communication regardless the network conditions. Therefore, can network assistance be used to improve the quality of OTT real-time communication services?To address this question, we study OTT solutions with a focus on WebRTC. We identify three loose coupling strategies that leverage network mechanisms for improving OTT communication services quality.We verify the pertinence of these coupling strategies in the context of traffic management. We identify two approaches of traffic management solutions adapted to WebRTC traffic: 1) aiming at assuring lower queuing delays regardless the traffic or 2) isolating the sensitive traffic. We study the impact of identified traffic management solutions on WebRTC for wireline access networks (uplink, ADSL and fiber). The obtained results show that current Internet engineering practices are not well adapted to the WebRTC traffic, but are optimized for TCP traffic. Furthermore, the proposed solutions ensure more fairness between WebRTC and TCP flows and consequently enable avoiding WebRTC traffic starvation and improve the overall quality of the communication.In the final analysis, the evaluated traffic management solutions are positioned in the context of identified coupling strategies. Based on this assessment, we provide recommendations of improving WebRTC quality with the assistance of NSP
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Schön, Martin. "Analýza závislostí komunikačních služeb na zpoždění a optimalizace QoS." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2015. http://www.nusl.cz/ntk/nusl-221343.
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This thesis discusses wireless network standards 802.11a/b/g/n. First part explains basic principles of networks and media access. Next the standard IEEE 802.11, general QoS parameters and their application in wireless networks, according to standard 802.11e are analyzed. Second part of the thesis verifies the acquired knowledge in simulating program Opnet - effects of the delay, jitter and packet loss on VoIP call are tested. In the last part of the thesis a network for video streaming has been designed. The video was streamed in different qualities and the influence of other network traffic (with and without the support of QoS) on the video streaming was tested.
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Bonfigli, Diego. "Supporto alla mobilità in sistemi multihomed eterogenei." Master's thesis, Alma Mater Studiorum - Università di Bologna, 2011. http://amslaurea.unibo.it/1914/.
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Contente, Pimentel Barbosa Douglas. "Análise de sistemas de telegonia IP em redes par-a-par sobrepostas." Universidade Federal de Pernambuco, 2008. https://repositorio.ufpe.br/handle/123456789/5227.
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Previous issue date: 2008Coordenação de Aperfeiçoamento de Pessoal de Nível Superior
As redes de telefonia IP popularizaram-se nos últimos anos sobretudo por seu baixo custo e facilidade de utilização. Transmitir voz na forma de pacotes IP favorece o desenvolvimento de uma rede integrada, na qual diversos tipos de dados e mídia trafegam segundo um padrão único, que uniformize os sistemas de telecomunicações (Convergência IP). As redes sobrepostas par-a-par são parcial ou totalmente independentes de qualquer servidor centralizado, possuem alta escalabilidade e fornecem meios para que a comunicação atravesse obstáculos impostos por NATs e firewalls. Tais redes oferecem aos pacotes uma maior flexibilidade de roteamento, permitindo que novas estratégias sejam utilizadas no encaminhamento dos pacotes. Essas estratégias proporcionam uma melhor qualidade de voz ao usuário, principalmente durante falhas e congestionamentos. Nesta dissertação são estudados os sistemas de comunicação de voz sobre IP (VoIP) arquitetados em topologias par-a-par sobrepostas. Aspectos de codificação, sinalização, roteamento e tráfego, bem como os protocolos envolvidos em tais sistemas são descritos. Alternativas para obter uma melhor qualidade de voz através do uso dessa configuração são analisadas. Como contribuições dessa dissertação, é realizada uma análise comparativa entre sistemas VoIP e apresentada uma nova forma quantitativa de medição da QoS (Qualidade de Serviço) baseada na correlação entre os sinais transmitidos e recebidos
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GOMES, Igor Ruiz. "Modelo de propagação indoor multi-andar em 2.4 GHz com estimativa de parâmetros de QoS em chamadas VoIP." Universidade Federal do Pará, 2010. http://repositorio.ufpa.br/jspui/handle/2011/2629.
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CNPq - Conselho Nacional de Desenvolvimento Científico e Tecnológico
O advento de novas formas multimídia tem atraído uma clientela exigente, onde preocupação não é somente com o serviço, mas também, com a qualidade que esse serviço pode ser oferecido. As WLAN (Wireless Local Area Networks) tornaram-se a forma mais comum de roteamento de Internet, devido ao seu baixo custo e facilidade de implementação. Para realizar um bom roteamento é necessário um planejamento, utilizando-se modelos. Os modelos de propagação existentes na literatura fazem a predição da intensidade do sinal, mas algumas vezes não contemplam a previsão de um bom serviço. Nesse sentido a presente dissertação propõe-se a elaborar um modelo de propagação empírico indoor multi-andar que não só prediz a potência recebida, mas também faz uma previsão para algumas métricas de QoS (Quality of Service) de chamadas VoIP (Voice over Internet Protocol). Para a elaboração do modelo proposto foram feitas campanhas de medição, em um prédio de dois andares, em pisos distintos mantendo-se a posição do ponto de acesso (PA) fixa. Estudos de geometria analítica para a contagem e agregação de perdas em pisos e paredes. Os resultados do modelo proposto foram comparados com um modelo da literatura que tem um comportamento similar, onde é possível verificar o melhor desempenho do modelo proposto, e para efeito de estudo um andar completamente simulado foi introduzido para avaliação.
The advent of new multimedia forms has attracted many customers, concerns not only with the service, but also with the quality of service that can be offered. The WLAN have become the most common form of Internet routing, this is because of its low cost and ease implementation. To achieve a good routing planning it’s necessary to use propagation models. In the literature many propagation models make the prediction of signal strength but do not include the provision of quality of service metrics (parameters). In this sense this work proposes to develop an empirical propagation model indoor multi-floor that not only predicts the received power, but also makes a prediction for some metrics of QoS for VoIP (Voice over Internet Protocol). To develop the proposed model, measurement campaigns were performed on separate floors of a building while maintaining the position of the access point AP) fixed in one floor. Studies of analytical geometry were taken for counting and aggregation of losses on floors/walls. The results of the proposed model were compared with model of literature which has similar propagation behavior. To improve a comparison, a test with a simulated floor was introduced using the proposed model.
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Díaz, Santos Juan Ramón. "Design and Implementation of a Communication Protocol to Improve Multimedia QoS and QoE in Wireless Ad Hoc Networks." Doctoral thesis, Universitat Politècnica de València, 2016. http://hdl.handle.net/10251/62162.
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[EN] This dissertation addresses the problem of multimedia delivery over multi-hop ad hoc wireless networks, and especially over wireless sensor networks. Due to their characteristics of low power consumption, low processing capacity and low memory capacity, they have major difficulties in achieving optimal quality levels demanded by end users in such communications.
In the first part of this work, it has been carried out a study to determine the behavior of a variety of multimedia streams and how they are affected by the network conditions when they are transmitted over topologies formed by devices of different technologies in multi hop wireless ad hoc mode. To achieve this goal, we have performed experimental tests using a test bench, which combine the main codecs used in audio and video streaming over IP networks with different sound and video captures representing the characteristic patterns of multimedia services such as phone calls, video communications, IPTV and video on demand (VOD). With the information gathered in the laboratory, we have been able to establish the correlation between the induced changes in the physical and logical topology and the network parameters that measure the quality of service (QoS) of a multimedia transmission, such as latency, jitter or packet loss. At this stage of the investigation, a study was performed to determine the state of the art of the proposed protocols, algorithms, and practical implementations that have been explicitly developed to optimize the multimedia transmission over wireless ad hoc networks, especially in ad hoc networks using clusters of nodes distributed over a geographic area and wireless sensor networks.
Next step of this research was the development of an algorithm focused on the logical organization of clusters formed by nodes capable of adapting to the circumstances of real-time traffic. The stated goal was to achieve the maximum utilization of the resources offered by the set of nodes that forms the network, allowing simultaneously sending reliably and efficiently all types of content through them, and mixing conventional IP data traffic with multimedia traffic with stringent QoS and QoE requirements. Using the information gathered in the previous phase, we have developed a network architecture that improves overall network performance and multimedia streaming. In parallel, it has been designed and programmed a communication protocol that allows implementing the proposal and testing its operation on real network infrastructures.
In the last phase of this thesis we have focused our work on sending multimedia in wireless sensor networks (WSN). Based on the above results, we have adapted both the architecture and the communication protocol for this particular type of network, whose use has been growing hugely in recent years.[ES] Esta tesis doctoral aborda el problema de la distribución de contenidos multimedia a través de redes inalámbricas ad hoc multisalto, especialmente las redes inalámbricas de sensores que, debido a sus características de bajo consumo energético, baja capacidad de procesamiento y baja capacidad de memoria, plantean grandes dificultades para alcanzar los niveles de calidad óptimos que exigen los usuarios finales en dicho tipo de comunicaciones. En la primera parte de este trabajo se ha llevado a cabo un estudio para determinar el comportamiento de una gran variedad de flujos multimedia y como se ven afectados por las condiciones de la red cuando son transmitidos a través topologías formadas por dispositivos de diferentes tecnologías que se comunican en modo ad hoc multisalto inalámbrico. Para ello, se han realizado pruebas experimentales sobre una maqueta de laboratorio, combinando los principales códecs empleados en la transmisión de audio y video a través de redes IP con diversas capturas de sonido y video que representan patrones característicos de servicios multimedia tales como las llamadas telefónicas, videoconferencias, IPTV o video bajo demanda (VOD). Con la información reunida en el laboratorio se ha podido establecer la correlación entre los cambios inducidos en la topología física y lógica de la red con los parámetros que miden la calidad de servicio (QoS) de una transmisión multimedia, tales como la latencia el jitter o la pérdida de paquetes. En esta fase de la investigación se realiza un estudio para determinar el estado del arte de las propuestas de desarrollo e implementación de protocolos y algoritmos que se han generado de forma explícita para optimizar la transmisión de tráfico multimedia sobre redes ad hoc inalámbricas, especialmente en las redes inalámbricas de sensores y redes ad hoc utilizando clústeres de nodos distribuidos en un espacio geográfico. El siguiente paso en la investigación ha consistido en el desarrollo de un algoritmo propio para la organización lógica de clústeres formados por nodos capaces de adaptarse a las circunstancias del tráfico en tiempo real. El objetivo planteado es conseguir un aprovechamiento máximo de los recursos ofrecidos por el conjunto de nodos que forman la red, permitiendo de forma simultánea el envío de todo tipo de contenidos a través de ellos de forma confiable y eficiente, permitiendo la convivencia de tráfico de datos IP convencional con tráfico multimedia con requisitos exigentes de QoS y QoE. A partir de la información conseguida en la fase anterior, se ha desarrollado una arquitectura de red que mejora el rendimiento general de la red y el de las transmisiones multimedia de audio y video en particular. De forma paralela, se ha diseñado y programado un protocolo de comunicación que permite implementar el modelo y testear su funcionamiento sobre infraestructuras de red reales. En la última fase de esta tesis se ha dirigido la atención hacia la transmisión multimedia en las redes de sensores inalámbricos (WSN). Partiendo de los resultados anteriores, se ha adaptado tanto la arquitectura como el protocolo de comunicaciones para este tipo concreto de red, cuyo uso se ha extendido en los últimos años de forma considerable
[CAT] Esta tesi doctoral aborda el problema de la distribució de continguts multimèdia a través de xarxes sense fil ad hoc multi salt, especialment les xarxes sense fil de sensors que, a causa de les seues característiques de baix consum energètic, baixa capacitat de processament i baixa capacitat de memòria, plantegen grans dificultats per a aconseguir els nivells de qualitat òptims que exigixen els usuaris finals en eixos tipus de comunicacions. En la primera part d'este treball s'ha dut a terme un estudi per a determinar el comportament d'una gran varietat de fluxos multimèdia i com es veuen afectats per les condicions de la xarxa quan són transmesos a través topologies formades per dispositius de diferents tecnologies que es comuniquen en mode ad hoc multi salt sense fil. Per a això, s'han realitzat proves experimentals sobre una maqueta de laboratori, combinant els principals códecs empleats en la transmissió d'àudio i vídeo a través de xarxes IP amb diverses captures de so i vídeo que representen patrons característics de serveis multimèdia com son les cridades telefòniques, videoconferències, IPTV o vídeo baix demanda (VOD). Amb la informació reunida en el laboratori s'ha pogut establir la correlació entre els canvis induïts en la topologia física i lògica de la xarxa amb els paràmetres que mesuren la qualitat de servei (QoS) d'una transmissió multimèdia, com la latència el jitter o la pèrdua de paquets. En esta fase de la investigació es realitza un estudi per a determinar l'estat de l'art de les propostes de desenvolupament i implementació de protocols i algoritmes que s'han generat de forma explícita per a optimitzar la transmissió de tràfic multimèdia sobre xarxes ad hoc sense fil, especialment en les xarxes sense fil de sensors and xarxes ad hoc utilitzant clusters de nodes distribuïts en un espai geogràfic. El següent pas en la investigació ha consistit en el desenvolupament d'un algoritme propi per a l'organització lògica de clusters formats per nodes capaços d'adaptar-se a les circumstàncies del tràfic en temps real. L'objectiu plantejat és aconseguir un aprofitament màxim dels recursos oferits pel conjunt de nodes que formen la xarxa, permetent de forma simultània l'enviament de qualsevol tipus de continguts a través d'ells de forma confiable i eficient, permetent la convivència de tràfic de dades IP convencional amb tràfic multimèdia amb requisits exigents de QoS i QoE. A partir de la informació aconseguida en la fase anterior, s'ha desenvolupat una arquitectura de xarxa que millora el rendiment general de la xarxa i el de les transmissions multimèdia d'àudio i vídeo en particular. De forma paral¿lela, s'ha dissenyat i programat un protocol de comunicació que permet implementar el model i testejar el seu funcionament sobre infraestructures de xarxa reals. En l'última fase d'esta tesi s'ha dirigit l'atenció cap a la transmissió multimèdia en les xarxes de sensors sense fil (WSN). Partint dels resultats anteriors, s'ha adaptat tant l'arquitectura com el protocol de comunicacions per a aquest tipus concret de xarxa, l'ús del qual s'ha estés en els últims anys de forma considerable.
Díaz Santos, JR. (2016). Design and Implementation of a Communication Protocol to Improve Multimedia QoS and QoE in Wireless Ad Hoc Networks [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/62162
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Martín, Severiano Juan Carlos. "IEEE 802.11b MAC layer's influence on VoIP quality parameters : Measurements and Analysis." Thesis, KTH, Mikroelektronik och informationsteknik, IMIT, 2004. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-92577.
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Real-time voice measurements were performed to assess whether there are significant problems with 802.11b wireless networks regarding real-time voice communication. We present an analysis of how the 802.11b MAC protocol and diverse environmental conditions affect the quality of real-time voice in terms of loss, delay, and jitter. We also reveal practical issues of wireless monitoring with passive sniffers for this type of analysis. The results obtained in our measurements show that in the majority of the experiments the quality was good, but under some circumstances the requirements for an acceptable voice communication were not met.Realtidsröstmätningar gjordes för att testa om det finns problem med 802.11b trådlösa nätverk beträffande realtidsröstkommunikation. En analys presenteras av hur 802.11b MACs protokoll och olika tillstånd i omgivningen påverkar kvaliteten på realtidsrösten i form av förluster, fördröjningar och jitter. Även praktiska angelägenheter om trådlös övervakning med passiva sniffers visas. De erhållna resultaten visar att i en majoritet av fallen var kvaliteten acceptabel, men under vissa förhållanden blev inte kraven för röstkommunikation uppfyllda.
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Ye, Dan. "Control of real-time multimedia applications in best-effort networks." [College Station, Tex. : Texas A&M University, 2006. http://hdl.handle.net/1969.1/ETD-TAMU-1157.
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Buchta, Marek. "Efektivita bezdrátových sítí z pohledu služeb." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2010. http://www.nusl.cz/ntk/nusl-218341.
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Subject of this thesis was the issue of effectiveness of wireless networks from the perspective of services. Become acquainted with WiFi network standard IEEE 802.11 with protocols and principles of WiFi networks and services. It should be focused on the quality of service QoS. In teoretical part of this thesis are described problems considering wireless networks, layer model and types of nets used in WiFi. Standard IEEE 802.11 is analysed in details including supplements and used packets. Special attention is paid to the quality of service QoS. Next are discussed the principles and application services as VoIP, videoconferencing, video streaming and others in WiFi network. On the basis of obtained knowledge is designed extensive WiFi network with assured supply of services. For modelling and simulation of wireless network is used software development environment Opnet Modeler. Model of wireless network is used to optimize and analyse this wireless communication network. In thesis are also simulated properties of data which are sensitive to delay. In last part of thesis is created a laboratory work, which contains a submission of task, detailed theory, wiring diagram, instructions for elaboration, guidelines for simulation and example of elaboration of task. Work is about comparison of network with used 802.11b and 802.11e standard, and support of quality of service QoS.
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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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Silva, Vandersilvio da. "Proposta de metodologia para avaliação de redes de voz sobre IP." reponame:Biblioteca Digital de Teses e Dissertações da UFRGS, 2006. http://hdl.handle.net/10183/8547.
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A redução de custo com telefonia através do uso de voz sobre IP tem disparado a busca de soluções que transformem redes IP originalmente dedicadas a transporte de dados em redes para transporte de voz. Esta dissertação tem por objetivo apresentar uma metodologia para sistematizar a avaliação de redes para o tráfego de voz sobre IP de acordo com as possibilidades disponíveis no cenário a ser avaliado. Inicialmente é dada uma visão geral de voz sobre IP, apresentando os protocolos utilizados, os fatores que influenciam na qualidade da voz e os métodos de avaliação de qualidade da voz. Na seqüência são apresentados trabalhos correlatos a avaliação de qualidade de aplicações de voz sobre IP. E por fim descreve-se a proposta de uma metodologia para sistematizar a avaliação de redes com VoIP.The use of voice over IP telephony was started with solutions to adapt existent data networks to carrier voice streams. The use of monitoring techniques, QoS and signaling protocols can be combined on a such design. Our goal is to present a methodology to evaluate and choose the probing points and the voice quality evaluation techniques to be used in network redesign. An overview about VoIP protocols and parameters that change the voice quality are presented as well as some related works on evaluating voice quality based on network parameters. A proposed methodology is presented, with a case study to show how one can choose the right combination of probing points with some voice quality measurement technique.
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Fredholm, Kenth, and Kristian Nilsson. "Implementing an application for communication and quality measurements over UMTS networks." Thesis, Linköping University, Department of Electrical Engineering, 2003. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-1666.
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The interest for various multimedia services accessed via the Internet has been growing immensely along with the bandwidth available. A similar development has emerged in the 3G mobile network. The focus of this master thesis is on the speech/audio part of a 3G multimedia application. The purpose has been to implement a traffic generating tool that can measure QoS (Quality of Service) in 3G networks. The application is compliant to the 3G standards, i.e. it uses AMR (Adaptive Multi Rate), SIP (Session Initiation Protocol) and RTP (Real Time Transport Protocol). AMR is a speech compression algorithm with the special feature that it can compress speech into several different bitrates. SIP signalling is used so that different applications can agree on how to communicate. RTP carries the speech frames over the network, in order to provide features that are necessary for media/multimedia applications. Issues like perception of audio and QoS related parameters is also discussed, from the perspective of users and developers.
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Kaman, Štěpán. "Implementace kvality služby v bezdrátových sítích." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2009. http://www.nusl.cz/ntk/nusl-218097.
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Wireless networks are currently the frequent access connection to the local network or the Internet. Until 2005 there was no support in the 802.11 standard priority data and the use of multimedia services in these networks is problematic. Thesis deals with the standard IEEE 802.11 standard and quality of service support including IEEE 802.11e. They discussed methods of access to transmission medium, differences in the MAC sublayer, the reader issues in the transmission of priority data and the requirements for these data. In Opnet Modeler was created wireless network with access points and stations on which they are carried out simulations at different strain of transmitted data. It studied the difference in the use of DCF and EDCF method, used in the network with QoS support. In particular, it examined the behavior of priority voice and video data in both networks. The focus is on key parameters such as throughput, dropping data, packet loss, delay, jitter and the size of broadcasting front. The measured data are analyzed, and differences in the network without the support and promoting the quality of services are compared and evaluated. Part of this work is the role of laboratory in the Opnet Modeler.
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Bachan, Jiří. "Metodika měření kvality služeb ve Wi-Fi sítích." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2008. http://www.nusl.cz/ntk/nusl-217542.
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This work deals with the wireless Wi-Fi networks, that used unlicenced ISM frequency band 2,4 GHz. These wireless networks are very extended in these days and used mainly for the Internet connection. With the multimedia data transfers expanding it’s necessary to ensure the specific quality of service QoS in wireless networks mainly for applications, which are sensitive to delay or packets lost. The main aim of this work is to describe Wi-Fi networks based on IEEE 802.11b/g/n standards, description of QoS techniques according to IEEE 802.11e standard and description of parameters describing QoS for the VoIP services. The practical part is divided on a two parts. The problems of measuring the radio link quality and the creation of simple program for quality of wireless connection analysis are solved in the first part. The second part incluledes measuring of the lost packets number in Wi-Fi network with the commercial application AirMagnet Laptop Wireless LAN Analyzer and measuring of the signal noise influence with the help of spectrum analyzer and vector signal generator.
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Mayer, Franz. "Adding NTP and RTCP to a SIP User Agent." Thesis, KTH, Kommunikationssystem, CoS, 2006. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-92200.
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With its enormous potential Voice over Internet Protocol is one of the latest buzzwords in information technology. Despite the numerous advantages of Voice over IP, it is a major technical challenge to achieve a similar call quality as experienced in the ordinary Public Switched Telephone Network. This thesis introduces standardized Internet protocols for Voice over IP, such as Session Initiation Protocol (SIP), Real-time Transport Protocol (RTP), in its background chapter. In order to provide better Quality of Service (QoS) Voice over IP applications should support a feedback mechanism, such as the Real-time Control Protocol (RTCP), and use accurate timing information, provided by the Network Time Protocol (NTP). Additionally this thesis considers synchronization issues in calls with two and more peers. After a rather academic overview of Voice over IP, the open source real-time application “minisip”, a SIP user agent, and its operation and structure for handling audio streams will be introduced. Minisip was extended by an implementation of NTP and RTCP to provide a test platform for this thesis. A clear conclusion is that the addition of global time helps facilitate synchronization of multiple streams from clients located any where in the network and in addition the ability to make one-way delay measurements helps SIP user agents to provide better quality audio to their users.Röst över IP, eller Internettelefoni baserad på “Voice over Internet Protocol” (VoIP), har med sin stora potential blivit ett av de senaste modeorden inom informationsteknologin. Vid sedan av ett antal fördelar med VoIP så innebär det en stor teknisk utmaning att uppnå en likadan samtalskvalitet som i det vanliga, fasta, telenätet. I den här uppsatsen beskrivs hur tjänstevalitet för VoIP kan förbättras genom att noggrant tidssynkronisera de (två eller flera) klienter som deltar i ett telefonsamtal. För detta krävs dels en återkopplingsmekanism, såsom “Real-time Control Protocol” (RTCP), samt en gemensam tidsuppfattning i de inblandade klienterna, vilket kan uppnås med hjälp av “Network Time Protocol” (NTP). Dessa protokoll, liksom de övriga Internet-standarder som VoIP baseras på (såsom “Session Initiation Protocol” (SIP) och “Real-time Transport Protocol” (RTP), beskrivs inledningsvis i uppsatsen. För studien har en SIP-klient baserad på öppen källkod använts (“Minisip”), och utökats med NTP och RCTP funktionalitet för att testa den föreslagna förbättringen av VoIP. En tydlig slutsats är att kännedom om en “global tid” möjliggör synkronisering av multipla ljudströmmar från klienter som befinner sig på olika nätverk. Möjligheten att mäta paketfördröjningen (envägs) bidrar också till en förbättrad ljudkvalitet.
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Oliveira, Eduardo Pinto de. "Ger?ncia de redes BWA atrav?s de Framework e m?tricas de avalia??o de desempenho." Pontif?cia Universidade Cat?lica de Campinas, 2009. http://tede.bibliotecadigital.puc-campinas.edu.br:8080/jspui/handle/tede/508.
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Previous issue date: 2009-02-13The objective of this work is to present the metrics of performance evaluation in Broadband Wireless Access (BWA). To achieve this goal it was necessary to build a Framework of network management customized to the equipment used. It was applied the knowledge in signals propagation of radio frequency (RF), radio planning of cellular systems, network management and statistical analyses. In the realization of the tests for the data collect it was configured a setup of tests with one Radio Base Station (BTS) and two Remote Stations (RS) located in the Pontif?cia Universidade Cat?lica de Campinas (PUC Campinas). It was used a technology commercially named as Pre-Wireless Interoperability of Multiple Accesses (Pre-WiMAX), operating in 5.8 GHz. This technology uses Orthogonal Frequency Division Multiplexing (OFDM) as multiplex technique and Carrier Sense Multiple Access (CSMA) as an access control to the media. In the development, it is presented a set of metrics related to the signals propagation by RF. The data collect was executed using the Simple Network Management Protocol (SNMP). The performance evaluation has considered the measures realized and its analytic and statistic treatments for a long time. By using the results of the measures, it was built the metrics that allow the evaluation of the performance in this network type, such as: link efficiency, throughput, bit error rate (BER), frame error rate (FER) and packet error rate (PER), among others. The work, as a whole, presents a strategy of practical evaluation applicable to BWA networks, creating indicators for decision-makings. To validate and complement the work it was done an Quality of Service (QoS) evaluation according to the International Telecommunication Union - Telecommunication sector (ITU-T) and the International Engineering Task Force (IETF) comparing the results with and without prioritization of Voice over Internet Protocol (VoIP) service .
O objetivo deste trabalho ? apresentar m?tricas de avalia??o de desempenho em Broadband Wireless Networks (BWA). Para atingir este objetivo foi necess?ria a constru??o de um Framework de ger?ncia de redes customizado aos equipamentos utilizados. Foram aplicados os conhecimentos de propaga??o de sinais de r?dio freq??ncia (RF), planejamento de sistemas celulares, gerenciamento de redes e an?lise estat?stica. Na realiza??o dos testes para a coleta de dados foi configurado um setup de testes com uma Esta??o R?dio Base (ERB) e duas Esta??es Remotas (ER), localizadas na Pontif?cia Universidade Cat?lica de Campinas (PUC Campinas). Para a valida??o da metodologia foi utilizada uma tecnologia denominada comercialmente como Pre-Wireless Interoperability of Multiple Accesses (Pre-WiMAX), operando em 5.8 GHz. Esta tecnologia utiliza Orthogonal Frequency Division Multiplexing (OFDM) como t?cnica de multiplexa??o e Carrier Sense Multiple Access (CSMA) como controle de acesso ao meio. No desenvolvimento ? apresentado um conjunto de m?tricas, referentes ? propaga??o de sinais por RF. A coleta de dados foi efetuada utilizando Simple Network Management Protocol (SNMP). A avalia??o de desempenho considerou as medidas realizadas e o seu tratamento anal?tico e estat?stico por longos per?odos de tempo. Com a utiliza??o dos resultados de medidas, foram constru?das as m?tricas que permitem avaliar o desempenho deste tipo de rede, tais como: efici?ncia de link, throughput, bit error rate (BER), frame error rate (FER), packet error rate (PER), entre outras. O trabalho como um todo apresenta uma estrat?gia de avalia??o pr?tica e aplic?vel a redes BWA, criando indicadores para tomadas de decis?o. Para validar e complementar o trabalho foi efetuada uma avalia??o de Quality of Service (QoS) segundo a International Telecommunication Union Telecommunication sector (ITU-T) e a International Engineering Task Force (IETF) com a compara??o de resultados sem e com prioriza??o do servi?o Voice over Internet Protocol (VoIP).
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Iqbal, Naveed, and Fahad-Mumtaz Cheema. "QoS_of_VoIP_in_Wireless_Networks." Thesis, Blekinge Tekniska Högskola, Avdelningen för telekommunikationssystem, 2009. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-1228.
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In this thesis we have focused in the wireless environment and how to run voice application over it. Conducive environment that makes it possible for the voice services to run in wireless is necessary. As we know this well that wireless is a contemporary technology due to it low cost and its effectiveness, and one major advantage of it is the mobility that is one fell free to move anywhere but have the access to the resource. So this makes wireless networks of great value, we in this thesis have focused on wireless LAN’s. In second part of the thesis we have shed some light on the VoIP showing how it works in the wireless environment. Analysis phase is relatively more important phase then the previous section which shows issues or hindrances in carrying voice over wireless environment. This analysis shows that these issues still prevails and should be addresses and the corresponding results are also discussed and by looking at those results we have derived a summery out of it. Next chapter we firstly tried to explain why we have chosen specific protocols and then showing some graphical representation measurements that are to address the problem based on the work done. We tried to evaluate EDCF and DCF as these play important role in handling real time applications like voice. After that we proposed a scheme through which these effects can be minimized and to enhance the method is necessary to avoid the issues still in effect.Thesis is Part for Master program in Electrical Engineering with Emphasis on Telecommunication(2007-2009).We have had a very nice time doing this thesis as there was alot of learning. Our examinator was allways there to help us, we are thankfull to Richard for his endless support.
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Mizera, Josef. "Podpora kvalitativních požadavků služeb v operačních systémech unixového typu pro provoz v bezdrátových sítích WiFi." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2014. http://www.nusl.cz/ntk/nusl-220637.
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Diploma thesis is focused on the supporting of Quality of Services in wireless networks, especially in the Linux operation systems. The topic is connected not only with OS, but also with the wireless standard, which supports QoS in wireless networks called IEEE 802.11e. QoS is needed especially for time-consuming data transfers in real time. The theoretical part deals with the theoretical analysis of the issue of the QoS support. There are described parameters, which occurred in quality of services support. This section also deals with the division of services that are used to transmit data across computer networks. It also describes the QoS support in wireless networks according 802.11e, its implementation and methods of accessing a medium with a without possibility of traffic. This part is followed by a description of QoS support in UNIX operating systems. The chapter describes how is the QoS support designed in these operating systems. There are also characterized concrete tools which are used for control the data flow in the operating systems using Linux. At the end the theoretical part deals with different types of queues and methods used in linux OS. In the practical part of the thesis, there are various designed topologies and scenarios to verify the functionality of QoS support in wireless networks using a Unix system. These chapters show the results of different tests at selected transmission data streams that are sensitive to transmission time. There is also verified cooperation of QoS support between devices operating on the network and data link layers. The output of this work is to design a laboratory exercise for the subject Network Architecture. This exercise is focused on familiarization with the QoS support functionality in wireless networks and in Unix-like operating systems. This chapter also describes the devices and programs that are needed to measure this task. The last part of the chapter describes the procedure for the preparation of the measuring station. For this laboratory task, there is an inserted manual in the annex.
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Novák, David. "Optimalizace přenosu hlasu v komunikačních sítích." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2010. http://www.nusl.cz/ntk/nusl-218340.
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This master’s thesis deals abou the transmission of voice in communications networks. The theoretical part describes criteria for optimizing voice, such as quality of service, type of service, level of service, service type, and mean opinion score. Next I describe the Internet Protocol, comparing IPv4 and IPv6, VoIP, including security, protocols and parameters necessary for transmission. Other part is about neural networks. There are basically described the neural network, Hopfield neural network and Kohenen neural network. The research is based on a comparison of the network without ensuring the quality of service and with ensuring quality of service. Then, there are compared two types of switches. Classical switch-controlled sequentially, and switch controlled by neural networks. The overall simulation program is implemented in Opnet Modeler. The conclusion deals with the creation of laboratory tasks in this program to compare the different systems of ensuring quality of service.
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Eriksson, Jhonny, and Joel Karlsson. "Granskning och optimering av data- och IP-telefoninätverk." Thesis, Mälardalen University, School of Innovation, Design and Engineering, 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:mdh:diva-9739.
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The company Västra Mälardalens Kommunalförbund, VMKF, wishes to revise and optimize their present data and IP-telephony network as of today consists of the three municipalities Köping, Arboga and Kungsör. As a municipal corporation, they seek consultation regarding internal as well as external review and investigation of the main structure of the network, its functionality and safety. By today’s increasing demands of Internet accessibility, availability of services and security far more extends the requirement of a complete network design. The foundation of networking rests on the balance between each of these necessities. Therefore, it is of grave importance to optimize a network design, use of hardware and to minimize the administrative overhead. In particular, when the municipality is short of resources and time means money. By letting an impartial investigation of the network act as a starting point it was established that several improvement could be applied. Among these a reconstructed and improved network topology that includes subjects as routing, switching, safety and security, quality of service and technical administrative overhead and the implementation of a real time monitoring of network bandwidth consumption.
Företaget Västra Mälardalens Kommunalförbund, VMKF, har önskemål om att granska och optimera deras befintliga data- och IP-telefoninätverk som i dagsläget spänner över de tre kommunerna Köping, Arboga och Kungsör. Som ett kommunalägt företag önskar de konsultation rörande intern såväl som extern granskning och optimering av huvuddelen av nätverkets funktionalitet samt säkerhet. I och med dagens ökade Internetanvändning och funktionalitetsbehov ställs allt högre krav på tillgänglighet, säkerhet och användarvänlighet. Nätverksteknik bygger mycket på balansen mellan dessa tre punkter. Därför gäller det att optimera nätverkets design, hårdvaruanvändning och att minimera administrativa laster. Detta i synnerhet då kommunens resurser är knappa och då tid i dagens samhälle innebär pengar. Genom att låta en granskning över nätverket som det ser ut i dag ligga till grund konstaterades att flertalet förbättringsmöjligheter kunde genomföras. Bland dessa återfinns en omstrukturerad nätverksdesign som innefattar routing, switching, säkerhet, QoS och teknisk administration samt implementeringen av en realtidsövervakning av bandbreddsanvändning.
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Rodrigues, Sandy Carmo Relva. "Encaminhamento óptimo do tráfego em redes Triple Play." Master's thesis, Universidade da Madeira, 2009. http://hdl.handle.net/10400.13/92.
Full textAbstract:
A Internet é responsável pelo surgimento de um novo paradigma de televisão – IPTV
(Televisão sobre IP). Este serviço distingue-se de outros modelos de televisão, pois permite aos utilizadores um elevado grau de interactividade, com um controlo personalizado sobre os conteúdos
a que pretende assistir. Possibilita ainda a oferta de um número ilimitado de canais, bem como o acesso a conteúdos de Vídeo on Demand (VoD). O IPTV apresenta diversas funcionalidades suportadas por uma arquitectura complexa e uma rede convergente que serve de integração a serviços de voz, dados e vídeo. A tecnologia IPTV explora ao máximo as características da Internet,
com a utilização de mecanismos de Qualidade de Serviço. Surge ainda como uma revolução dentro do panorama televisivo, abrindo portas a novos investimentos por parte das empresas de telecomunicações. A Internet também permite fazer chamadas telefónicas sobre a rede IP. Este serviço é denominado VoIP (Voz sobre IP) e encontra-se em funcionamento já há algum tempo.
Desta forma surge a oportunidade de poder oferecer ao consumidor final, um serviço que
inclua os serviços de Internet, de VoIP e de IPTV denominado serviço Triple Play. O serviço Triple Play veio obrigar a revisão de toda a rede de transporte de forma a preparar a mesma para suportar este serviço de uma forma eficiente (QoS), resiliente (recuperação de falhas) e optimizado
(Engenharia de tráfego).
Em redes de telecomunicações, tanto a quebra de uma ligação como a congestão nas redes
pode interferir nos serviços oferecidos aos consumidores finais. Mecanismos de sobrevivência são aplicados de forma a garantir a continuidade do serviço mesmo na ocorrência de uma falha.
O objectivo desta dissertação é propor uma solução de uma arquitectura de rede capaz de
suportar o serviço Triple Play de uma forma eficiente, resiliente e optimizada através de um encaminhamento óptimo ou quase óptimo. No âmbito deste trabalho, é realizada a análise do impacto das estratégias de encaminhamento que garantem a eficiência, sobrevivência e optimização das redes IP existentes, bem como é determinado o número limite de clientes permitido numa situação de pico de uma dada rede.
Neste trabalho foram abordados os conceitos de Serviços Triple Play, Redes de Acesso,
Redes Núcleo, Qualidade de Serviço, MPLS (Multi-Protocolo Label Switching), Engenharia de Tráfego e Recuperação de falhas. As conclusões obtidas das simulações efectuadas através do simulador de rede NS-2.33 (Network Simulator versão 2.33) serviram para propor a solução da arquitectura de uma rede capaz de suportar o serviço Triple Play de uma forma eficiente, resiliente e optimizada.Orientador: Paulo Nazareno Maia Sampaio
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Nivor, Frédéric. "Architecture de Communication pour les Applications Multimédia Interactives dans les Réseaux Sans Fil." Phd thesis, Université Paul Sabatier - Toulouse III, 2009. http://tel.archives-ouvertes.fr/tel-01067146.
Full textAbstract:
Les travaux de cette thèse s'inscrivent dans le contexte des réseaux sans fil et des réseaux d'accès par Satellite en particulier, qui facilitent l'installation d'infrastructures réseau dans les zones géographiquement reculées et faiblement peuplées. Cependant, ces derniers présentent certains inconvénients lorsqu'il s'agit de déployer des applications multimédia interactives. En effet, de telles applications requièrent un délai de bout en bout aussi faible que possible et plus généralement exigent une meilleure Qualité de Service (QdS) du système de communication que le classique Meilleur-Effort (BE) afin de fonctionner correctement. Or, les réseaux d'accès par satellite géostationnaires souffrent déjà d'un délai de propagation non négligeable d'autant plus accru que la transmission des données est assurée par des mécanismes d'allocation dynamique, par exemple dans un système DVB-S2/RCS. Dans ces travaux de thèse, nous proposons d'utiliser les informations de signalisation de session des applications multimédia basées sur le protocole de session SIP afin d'ajuster le paramétrage du système de communication selon une approche " cross-layer " qui permet alors d'améliorer de façon significative la réactivité du système. Nous avons proposé plusieurs solutions pour, d'abord réduire le temps entre la demande de communication et le démarrage effectif du transfert des flux multimédia, ensuite réduire le délai de transmission des données multimédia durant la communication (tout en utilisant de manière optimale les ressources réseau disponibles sur la voie retour), et enfin accroître le nombre de flux multimédia admissibles dans le réseau satellite tout en leur garantissant un niveau de QdS satisfaisant. Afin de faciliter l'intégration et l'implémentation des solutions proposées dans un système de communication réel, un mécanisme de communication inter-couches d'optimisation est proposé et développé. De plus, une architecture orientée web services est utilisée afin de faciliter la découverte et l'invocation des différentes niveaux de services de communication présents dans de tels réseaux d'accès. Les solutions proposées ont été évaluées dans des environnements sans fil émulés et réels