SSthreshless Start: A Sender-Side TCP Intelligence for Long Fat Network

Measurement shows that 85% of TCP flows in the internet are short-lived flows that stay most of their operation in the TCP startup phase. However, many previous studies indicate that the traditional TCP Slow Start algorithm does not perform well, especially in long fat networks. Two obvious problems are known to impact the Slow Start performance, which are the blind initial setting of the Slow Start threshold and the aggressive increase of the probing rate during the startup phase regardless of the buffer sizes along the path. Current efforts focusing on tuning the Slow Start threshold and/or probing rate during the startup phase have not been considered very effective, which has prompted an investigation with a different approach. In this paper, we present a novel TCP startup method, called threshold-less slow start or SSthreshless Start, which does not need the Slow Start threshold to operate. Instead, SSthreshless Start uses the backlog status at bottleneck buffer to adaptively adjust probing rate which allows better seizing of the available bandwidth. Comparing to the traditional and other major modified startup methods, our simulation results show that SSthreshless Start achieves significant performance improvement during the startup phase. Moreover, SSthreshless Start scales well with a wide range of buffer size, propagation delay and network bandwidth. Besides, it shows excellent friendliness when operating simultaneously with the currently popular TCP NewReno connections.Comment: 25 pages, 10 figures, 7 table

An Energy and Performance Exploration of Network-on-Chip Architectures

In this paper, we explore the designs of a circuit-switched router, a wormhole router, a quality-of-service (QoS) supporting virtual channel router and a speculative virtual channel router and accurately evaluate the energy-performance tradeoffs they offer. Power results from the designs placed and routed in a 90-nm CMOS process show that all the architectures dissipate significant idle state power. The additional energy required to route a packet through the router is then shown to be dominated by the data path. This leads to the key result that, if this trend continues, the use of more elaborate control can be justified and will not be immediately limited by the energy budget. A performance analysis also shows that dynamic resource allocation leads to the lowest network latencies, while static allocation may be used to meet QoS goals. Combining the power and performance figures then allows an energy-latency product to be calculated to judge the efficiency of each of the networks. The speculative virtual channel router was shown to have a very similar efficiency to the wormhole router, while providing a better performance, supporting its use for general purpose designs. Finally, area metrics are also presented to allow a comparison of implementation costs

VoIP: Making Secure Calls and Maintaining High Call Quality

Modern multimedia communication tools must have high security, high availability and high quality of service (QoS). Any security implementation will directly impact on QoS. This paper will investigate how end-to-end security impacts on QoS in Voice over Internet Protocol (VoIP). The QoS is measured in terms of lost packet ratio, latency and jitter using different encryption algorithms, no security and just the use of IP firewalls in Local and Wide Area Networks (LAN and WAN). The results of laboratory tests indicate that the impact on the overall performance of VoIP depends upon the bandwidth availability and encryption algorithm used. The implementation of any encryption algorithm in low bandwidth environments degrades the voice quality due to increased loss packets and packet latency, but as bandwidth increases encrypted VoIP calls provided better service compared to an unsecured environment.Les eines modernes de comunicació multimèdia han de tenir alta seguretat, alta disponibilitat i alta qualitat de servei (QoS). Cap tipus d¿implementació de seguretat tindrà un impacte directe en la qualitat de servei. En aquest article s¿investiga com la seguretat d'extrem a extrem impacta en la qualitat de servei de veu sobre el Protocol d'Internet (VoIP). La qualitat de servei es mesura en termes de pèrdua de proporció de paquets, latència i jitter utilitzant diferents algoritmes d¿encriptació, sense seguretat i només amb l'ús de tallafocs IP en local i en xarxes d'àrea àmplia (LAN i WAN). Els resultats de les proves de laboratori indiquen que l'impacte general sobre el rendiment de VoIP depèn de la disponibilitat d'ample de banda i l'algorisme de xifrat que s'utilitza. La implementació de qualsevol algorisme de xifrat en entorns de baix ample de banda degrada la veu a causa de l'augment de la pèrdua de paquets i latència dels paquets de qualitat, però quan l'ample de banda augmenta les trucades de VoIP xifrades proporcionen un millor servei en comparació amb un entorn sense seguretat.Las herramientas modernas de comunicación multimedia deben tener alta seguridad, alta disponibilidad y alta calidad de servicio (QoS). Ningún tipo de implementación de seguridad tendrá un impacto directo en la calidad de servicio. En este artículo se investiga como la seguridad de extremo a extremo impacta en la calidad de servicio de voz sobre el Protocolo de Internet (VoIP). La calidad de servicio se mide en términos de pérdida de proporción de paquetes, latencia y jitter utilizando diferentes algoritmos de encriptación, sin seguridad y sólo con el uso de cortafuegos IP en local y en redes de área amplia (LAN y WAN). Los resultados de las pruebas de laboratorio indican que el impacto general sobre el rendimiento de VoIP depende de la disponibilidad de ancho de banda y el algoritmo de cifrado que se utiliza. La implementación de cualquier algoritmo de cifrado en entornos de bajo ancho de banda degrada la voz debido al aumento de la pérdida de paquetes y latencia de los paquetes de calidad, pero cuando el ancho de banda aumenta las llamadas de VoIP cifradas proporcionan un mejor servicio en comparación con un entorno sin seguridad

TCP smart framing: a segmentation algorithm to reduce TCP latency

TCP Smart Framing, or TCP-SF for short, enables the Fast Retransmit/Recovery algorithms even when the congestion window is small. Without modifying the TCP congestion control based on the additive-increase/multiplicative-decrease paradigm, TCP-SF adopts a novel segmentation algorithm: while Classic TCP always tries to send full-sized segments, a TCP-SF source adopts a more flexible segmentation algorithm to try and always have a number of in-flight segments larger than 3 so as to enable Fast Recovery. We motivate this choice by real traffic measurements, which indicate that today's traffic is populated by short-lived flows, whose only means to recover from a packet loss is by triggering a Retransmission Timeout. The key idea of TCP-SF can be implemented on top of any TCP flavor, from Tahoe to SACK, and requires modifications to the server TCP stack only, and can be easily coupled with recent TCP enhancements. The performance of the proposed TCP modification were studied by means of simulations, live measurements and an analytical model. In addition, the analytical model we have devised has a general scope, making it a valid tool for TCP performance evaluation in the small window region. Improvements are remarkable under several buffer management schemes, and maximized by byte-oriented schemes

Multiclass scheduling algorithms for the DAVID metro network

Abstract—The data and voice integration over dense wavelength-division-multiplexing (DAVID) project proposes a metro network architecture based on several wavelength-division-multiplexing (WDM) rings interconnected via a bufferless optical switch called Hub. The Hub provides a programmable interconnection among rings on the basis of the outcome of a scheduling algorithm. Nodes connected to rings groom traffic from Internet protocol routers and Ethernet switches and share ring resources. In this paper, we address the problem of designing efficient centralized scheduling algorithms for supporting multiclass traffic services in the DAVID metro network. Two traffic classes are considered: a best-effort class, and a high-priority class with bandwidth guarantees. We define the multiclass scheduling problem at the Hub considering two different node architectures: a simpler one that relies on a complete separation between transmission and reception resources (i.e., WDM channels) and a more complex one in which nodes fully share transmission and reception channels using an erasure stage to drop received packets, thereby allowing wavelength reuse. We propose both optimum and heuristic solutions, and evaluate their performance by simulation, showing that heuristic solutions exhibit a behavior very close to the optimum solution. Index Terms—Data and voice integration over dense wavelength-division multiplexing (DAVID), metropolitan area network, multiclass scheduling, optical ring, wavelength-division multiplexing (WDM). I