Small-Packet Flows in Software Defined Networks: Traffic Profile Optimization

This paper proposes a method for optimizing bandwidth usage in Software Defined Networks (SDNs) based on OpenFlow. Flows of small packets presenting a high overhead, as the ones generated by emerging services, can be identified by the SDN controller, in order to remove header fields that are common to any packet in the flow, only during their way through the SDN. At the same time, several packets can be multiplexed together in the same frame, thus reducing the overall number of frames. The method can be useful for providing QoS while the packets are traversing the SDN. Four kinds of small-packet traffic flows are considered (VoIP, UDP and TCP-based online games, and ACKs from TCP flows). Both IPv4 and IPv6 are studied, and significant bandwidth savings (up to 68 % for IPv4 and 78 % for IPv6) can be obtained for the considered kinds of traffic. The optimization method is also applied to different public Internet traffic traces, and significant reductions in terms of packets per second are achieved. Results show that bandwidth consumption is also reduced, especially in those traces where the percentage of small packets is high. Regarding the effect on QoS, the additional delay can be kept very low (below 1 millisecond) when the throughput is high, but it may become significant for low- throughput scenarios. Thus, a trade-off between bandwidth saving and additional delay appears in those cases

Optimization of low-efficiency traffic in OpenFlow Software Defined Networks

Abstract — This paper proposes a method for optimizing bandwidth usage in Software Defined Networks (SDNs) based on OpenFlow. Flows of small packets presenting a high overhead, as the ones generated by emerging services, can be identified by the SDN controller, in order to remove header fields that are common to any packet in the flow, only during their way through the SDN. At the same time, several packets can be multiplexed together in the same frame, thus reducing the number of sent frames. Four kinds of small-packet traffic flows are considered (VoIP, UDP and TCP-based online games, and ACKs from TCP flows). Both IPv4 and IPv6 are tested, and significant bandwidth savings (up to 68 % for IPv4 and 78 % for IPv6) can be obtained for the considered kinds of traffic

The Effect of the Buffer Size in QoS for Multimedia and bursty Traffic: When an Upgrade Becomes a Downgrade

This work presents an analysis of the buffer features of an access router, especially the size, the impact on delay and the packet loss rate. In particular, we study how these features can affect the Quality of Service (QoS) of multimedia applications when generating traffic bursts in local networks. First, we show how in a typical SME (Small and Medium Enterprise) network in which several multimedia flows (VoIP, videoconferencing and video surveillance) share access, the upgrade of the bandwidth of the internal network may cause the appearance of a significant amount of packet loss caused by buffer overflow. Secondly, the study shows that the bursty nature of the traffic in some applications traffic (video surveillance) may impair their QoS and that of other services (VoIP and videoconferencing), especially when a certain number of bursts overlap. Various tests have been developed with the aim of characterizing the problems that may appear when network capacity is increased in these scenarios. In some cases, especially when applications generating bursty traffic are present, increasing the network speed may lead to a deterioration in the quality. It has been found that the cause of this quality degradation is buffer overflow, which depends on the bandwidth relationship between the access and the internal networks. Besides, it has been necessary to describe the packet loss distribution by means of a histogram since, although most of the communications present good QoS results, a few of them have worse outcomes. Finally, in order to complete the study we present the MOS results for VoIP calculated from the delay and packet loss rate

Quality aspects of Internet telephony

Internet telephony has had a tremendous impact on how people communicate. Many now maintain contact using some form of Internet telephony. Therefore the motivation for this work has been to address the quality aspects of real-world Internet telephony for both fixed and wireless telecommunication. The focus has been on the quality aspects of voice communication, since poor quality leads often to user dissatisfaction. The scope of the work has been broad in order to address the main factors within IP-based voice communication. The first four chapters of this dissertation constitute the background material. The first chapter outlines where Internet telephony is deployed today. It also motivates the topics and techniques used in this research. The second chapter provides the background on Internet telephony including signalling, speech coding and voice Internetworking. The third chapter focuses solely on quality measures for packetised voice systems and finally the fourth chapter is devoted to the history of voice research. The appendix of this dissertation constitutes the research contributions. It includes an examination of the access network, focusing on how calls are multiplexed in wired and wireless systems. Subsequently in the wireless case, we consider how to handover calls from 802.11 networks to the cellular infrastructure. We then consider the Internet backbone where most of our work is devoted to measurements specifically for Internet telephony. The applications of these measurements have been estimating telephony arrival processes, measuring call quality, and quantifying the trend in Internet telephony quality over several years. We also consider the end systems, since they are responsible for reconstructing a voice stream given loss and delay constraints. Finally we estimate voice quality using the ITU proposal PESQ and the packet loss process. The main contribution of this work is a systematic examination of Internet telephony. We describe several methods to enable adaptable solutions for maintaining consistent voice quality. We have also found that relatively small technical changes can lead to substantial user quality improvements. A second contribution of this work is a suite of software tools designed to ascertain voice quality in IP networks. Some of these tools are in use within commercial systems today

On the effectiveness of an optimization method for the traffic of TCP-based multiplayer online games

This paper studies the feasibility of using an optimization method, based on multiplexing and header compression, for the traffic of Massively Multiplayer Online Role Playing Games (MMORPGs) using TCP at the Transport Layer. Different scenarios where a number of flows share a common network path are identified. The adaptation of the multiplexing method is explained, and a formula of the savings is devised. The header compression ratio is obtained using real traces of a popular game and a statistical model of its traffic is used to obtain the bandwidth saving as a function of the number of players and the multiplexing period. The obtained savings can be up to 60 % for IPv4 and 70 % for IPv6. A Mean Opinion Score model from the literature is employed to calculate the limits of the multiplexing period that can be used without harming the user experience. The interactions between multiplexed and non-multiplexed flows, sharing a bottleneck with different kinds of background traffic, are studied through simulations. As a result of the tests, some limits for the multiplexing period are recommended: the unfairness between players can be low if the value of the multiplexing period is kept under 10 or 20 ms. TCP background flows using SACK (Selective Acknowledgment) and Reno yield better results, in terms of fairness, than Tahoe and New Reno. When UDP is used for background traffic, high values of the multiplexing period may stress the unfairness between flows if network congestion is severe

Enhancement of perceived quality of service for voice over internet protocol systems

Voice over Internet Protocol (WIP) applications are becoming more and more popular in the telecommunication market. Packet switched V61P systems have many technical advantages over conventional Public Switched Telephone Network (PSTN), including its efficient and flexible use of the bandwidth, lower cost and enhanced security. However, due to the IP network's "Best Effort" nature, voice quality are not naturally guaranteed in the VoIP services. In fact, most current Vol]P services can not provide as good a voice quality as PSTN. IP Network impairments such as packet loss, delay and jitter affect perceived speech quality as do application layer impairment factors, such as codec rate and audio features. Current perceived Quality of Service (QoS) methods are mainly designed to be used in a PSTN/TDM environment and their performance in V6IP environment is unknown. It is a challenge to measure perceived speech quality correctly in V61P system and to enhance user perceived speech quality for VoIP system. The main goal of this project is to evaluate the accuracy of the existing ITU-T speech quality measurement method (Perceptual Evaluation of Speech Quality - PESQ) in mobile wireless systems in the context of V61P, and to develop novel and efficient methods to enhance the user perceived speech quality for emerging V61P services especially in mobile V61P environment. The main contributions of the thesis are threefold: (1) A new discovery of PESQ errors in mobile VoIP environment. A detailed investigation of PESQ performance in mobile VoIP environment was undertaken and included setting up a PESQ performance evaluation platform and testing over 1800 mobile-to-mobile and mobileto- PSTN calls over a period of three months. The accuracy issues of PESQ algorithm was investigated and main problems causing inaccurate PESQ score (improper time-alignment in the PESQ algorithm) were discovered . Calibration issues for a safe and proper PESQ testing in mobile environment were also discussed in the thesis. (2) A new, simple-to-use, V611Pjit ter buffer algorithm. This was developed and implemented in a commercial mobile handset. The algorithm, called "Play Late Algorithm", adaptively alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. It can be used as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance. Results show that the proposed algorithm can increase user perceived quality without consuming too much processing power when tested in live wireless VbIP networks. (3) A new QoS enhancement scheme. The new scheme combines the strengths of adaptive codec bit rate (i. e. AMR 8-modes bit rate) and speech priority marking (i. e. giving high priority for the beginning of a voiced segment). The results gathered on a simulation and emulation test platform shows that the combined method provides a better user perceived speech quality than separate adaptive sender bit rate or packet priority marking methods

Final report on the evaluation of RRM/CRRM algorithms

Deliverable public del projecte EVERESTThis deliverable provides a definition and a complete evaluation of the RRM/CRRM algorithms selected in D11 and D15, and evolved and refined on an iterative process. The evaluation will be carried out by means of simulations using the simulators provided at D07, and D14.Preprin

Quality of service optimization of multimedia traffic in mobile networks

Mobile communication systems have continued to evolve beyond the currently deployed Third Generation (3G) systems with the main goal of providing higher capacity. Systems beyond 3G are expected to cater for a wide variety of services such as speech, data, image transmission, video, as well as multimedia services consisting of a combination of these. With the air interface being the bottleneck in mobile networks, recent enhancing technologies such as the High Speed Downlink Packet Access (HSDPA), incorporate major changes to the radio access segment of 3G Universal Mobile Telecommunications System (UMTS). HSDPA introduces new features such as fast link adaptation mechanisms, fast packet scheduling, and physical layer retransmissions in the base stations, necessitating buffering of data at the air interface which presents a bottleneck to end-to-end communication. Hence, in order to provide end-to-end Quality of Service (QoS) guarantees to multimedia services in wireless networks such as HSDPA, efficient buffer management schemes are required at the air interface. The main objective of this thesis is to propose and evaluate solutions that will address the QoS optimization of multimedia traffic at the radio link interface of HSDPA systems. In the thesis, a novel queuing system known as the Time-Space Priority (TSP) scheme is proposed for multimedia traffic QoS control. TSP provides customized preferential treatment to the constituent flows in the multimedia traffic to suit their diverse QoS requirements. With TSP queuing, the real-time component of the multimedia traffic, being delay sensitive and loss tolerant, is given transmission priority; while the non-real-time component, being loss sensitive and delay tolerant, enjoys space priority. Hence, based on the TSP queuing paradigm, new buffer managementalgorithms are designed for joint QoS control of the diverse components in a multimedia session of the same HSDPA user. In the thesis, a TSP based buffer management algorithm known as the Enhanced Time Space Priority (E-TSP) is proposed for HSDPA. E-TSP incorporates flow control mechanisms to mitigate congestion in the air interface buffer of a user with multimedia session comprising real-time and non-real-time flows. Thus, E-TSP is designed to provide efficient network and radio resource utilization to improve end-to-end multimedia traffic performance. In order to allow real-time optimization of the QoS control between the real-time and non-real-time flows of the HSDPA multimedia session, another TSP based buffer management algorithm known as the Dynamic Time Space Priority (D-TSP) is proposed. D-TSP incorporates dynamic priority switching between the real-time and non-real-time flows. D-TSP is designed to allow optimum QoS trade-off between the flows whilst still guaranteeing the stringent real-time component’s QoS requirements. The thesis presents results of extensive performance studies undertaken via analytical modelling and dynamic network-level HSDPA simulations demonstrating the effectiveness of the proposed TSP queuing system and the TSP based buffer management schemes

Energy efficiency in next generation wireless networks: methodologies, solutions and algorithms

Mobile Broadband Wireless Access (BWA) networks will offer in the forthcoming years multiple and differentiated services to users with high mobility requirements, connecting via portable or wearable devices which rely on the use of batteries by necessity. Since such devices consume a relatively large fraction of energy for transmitting/receiving data over-the-air, mechanisms are needed to reduce power consumption, in order to increase the lifetime of devices and hence improve user’s satisfaction. Next generation wireless network standards define power saving functions at the Medium Access Control (MAC) layer, which allow user terminals to switch off the radio transceiver during open traffic sessions for greatest energy consumption reduction. However, enabling power saving usually increases the transmission latency, which can negatively affect the Quality of Service (QoS) experienced by users. On the other hand, imposing stringent QoS requirements may limit the amount of energy that can be saved. The IEEE 802.16e standard defines the sleep mode is power saving mechanism with the purpose of reducing energy consumption. Three different operation classes are provided, each one to serve different class of traffic: class I, best effort traffic, class II real time traffic and class III multicast traffic. Several aspects of the sleep mode are left unspecified, as it is usually done in standards, allowing manufacturers to implement their own proprietary solutions, thus gaining a competitive advantage over the rivals. The work of this thesis is aimed at verifying, the effectiveness of the power saving mechanism proposed into IEEE 802.16e standard, focusing on the mutual interaction between power saving and QoS support. Two types of delay constrained applications with different requirements are considered, i.e., Web and Voice over IP (VoIP). The performance is assessed via detailed packet-level simulation, with respect to several system parameters. To capture the relative contribution of all the factors on the energy- and QoS-related metrics, part of the evaluation is carried out by means of 2k · r! analysis. Our study shows that the sleep mode can achieve significant power consumption reduction, however, when real time traffic is considered a wise configuration of the parameters is mandatory in order to avoid unacceptable degradation of the QoS. Finally, based on the guidelines drawn through the analysis, we extend our contribution beyond a simple evaluation, proposing a power saving aware scheduling framework aimed at reducing further the energy consumption. Our framework integrates with existing scheduling policies that can pursue their original goals, e.g. maximizing throughput or fairness, while improving the energy efficiency of the user terminals. Its effectiveness is assessed through an extensive packet level simulation campaign