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  • 1.
    Abrahamsson, Henrik
    et al.
    RISE, Swedish ICT, SICS, Decisions, Networks and Analytics lab.
    Hagsand, Olof
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    TCP over High Speed Variable Capacity Links: A Simulation Study for Bandwidth Allocation2002Conference paper (Refereed)
  • 2.
    Abrahamsson, Henrik
    et al.
    RISE, Swedish ICT, SICS, Decisions, Networks and Analytics lab.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    DTMsim - DTM channel simulation in ns2001Report (Other academic)
    Abstract [en]

    Dynamic Transfer Mode (DTM) is a ring based MAN technology that provides a channel abstraction with a dynamically adjustable capacity. TCP is a reliable end to end transport protocol capable of adjusting its rate. The primary goal of this work is investigate the coupling of dynamically allocating bandwidth to TCP flows with the affect this has on the congestion control mechanism of TCP. In particular we wanted to find scenerios where this scheme does not work, where either all the link capacity is allocated to TCP or congestion collapse occurs and no capacity is allocated to TCP. We have created a simulation environment using ns-2 to investigate TCP over networks which have a variable capacity link. We begin with a single TCP Tahoe flow over a fixed bandwidth link and progressively add more complexity to understand the behaviour of dynamically adjusting link capacity to TCP and vice versa.

  • 3.
    Ahlgren, Bengt
    et al.
    RISE, Swedish ICT, SICS, Decisions, Networks and Analytics lab.
    Andersson, Anders
    Hagsand, Olof
    RISE - Research Institutes of Sweden, ICT, SICS.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Dimensioning links for IP telephony2001In: Proceedings of the 2nd IP-Telephony Workshop (IPtel 2001), 2-3 April 2001, New York City, New York, USA, 2001, 1Conference paper (Refereed)
    Abstract [en]

    Packet loss is an important parameter for dimensioning network links or traffic classes carrying IP telephony traffic. We present a model based on the Markov modulated Poisson process (MMPP) which calculates packet loss probabilities for a set of super positioned voice input sources and the specified link properties. We do not introduce another new model to the community, rather try and verify one of the existing models via extensive simulation and a real world implementation. A plethora of excellent research on queuing theory is still in the domain of ATM researchers and we attempt to highlight its validity to the IP Telephony community. Packet level simulations show very good correspondence with the predictions of the model. Our main contribution is the verification of the MMPP model with measurements in a laboratory environment. The loss rates predicted by the model are in general close to the measured loss rates and the loss rates obtained with simulation. The general conclusion is that the MMPP-based model is a tool well suited for dimensioning links carrying packetized voice in a system with limited buffer space.

  • 4.
    Ahlgren, Bengt
    et al.
    RISE, Swedish ICT, SICS, Decisions, Networks and Analytics lab.
    Andersson, Anders
    RISE - Research Institutes of Sweden, ICT, SICS.
    Hagsand, Olof
    RISE - Research Institutes of Sweden, ICT, SICS.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Dimensioning Links for IP Telephony2000Report (Other academic)
    Abstract [en]

    Transmitting telephone calls over the Internet causes problems not present in current telephone technology such as packet loss and delay due to queueing in routers. In this undergraduate thesis we study how a Markov modulated Poisson process is applied as an arrival process to a multiplexer and we study the performance in terms of loss probability. The input consists of the superposition of independent voice sources. The predictions of the model is compared with results obtained with simulations of the multiplexer made with a network simulator. The buffer occupancy distribution is also studied and we see how this distribution changes as the load increases.

  • 5.
    Ahlgren, Bengt
    et al.
    RISE, Swedish ICT, SICS, Decisions, Networks and Analytics lab.
    D’Ambrosio, Matteo
    4WARD.
    Dannewitz, Christian
    4WARD.
    Marchisio, Marco
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Ohlman, Börje
    Pentikousis, Kostas
    Rembarz, René
    Strandberg, Ove
    Vercellone, Vinicio
    Design considerations for a network of information2008Conference paper (Refereed)
    Abstract [en]

    The existing Internet ecosystem is a result of decades of evolution. It has managed to scale well beyond the original aspirations. Evolution, though, highlighted a certain degree of inadequacies that is well documented. In this position paper we present the design considerations for a re-architected global networking architecture which delivers dissemination and non-dissemination objects only to consenting recipients, reducing unwanted traffic, linking information producers with consumers independently of the hosts involved, and connects the digital with the physical world. We consider issues ranging from the proposed object identifier/locator split to security and trust as we transition towards a Network of Information and relate our work with the emerging paradigm of publish/subscribe architectures. We introduce the fundamental components of a Network of Information, i.e., name resolution, routing, storage, and search, and close this paper with a discussion about future work.

  • 6. Biyani, Pravesh
    et al.
    Hagsand, Olof
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Early estimation of voip quality2003Conference paper (Refereed)
  • 7.
    Bjurling, Björn
    et al.
    RISE, Swedish ICT, SICS, Decisions, Networks and Analytics lab.
    Kreuger, Per
    RISE, Swedish ICT, SICS, Decisions, Networks and Analytics lab.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Data Readiness for BADA: BADA main study 1, FFI/Vinnova grant 2015-006772017Report (Other academic)
  • 8.
    Filippini, Ilario
    et al.
    CNS.
    Malandrino, Francesco
    CNS.
    Dan, Gyorgy
    CNS.
    Cesana, Matteo
    CNS.
    Casetti, Claudio
    CNS.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Non-cooperative RSU deployment in vehicular networks2011Conference paper (Refereed)
    Abstract [en]

    This work considers the issue of distributing con- tents to vehicles through roadside communication infrastructure. Within this scenario, this work studies the dynamics of infras- tructure deployment by using game theoretic tools. A strategic game is used to model the case in which the operators perform their deployment decisions concurrently, whereas an extensive game is used to study the dynamics in case one operator is the deployment leader and moves first. The equilibria of the aforementioned games are then assessed as a function of several parameters (nominal infrastructure capacity, interference, vehicle flows). Simulations are used to validate the analytical findings.

  • 9.
    Grönvall, Björn
    et al.
    RISE - Research Institutes of Sweden, ICT, SICS.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Pink, Stephen
    RISE - Research Institutes of Sweden, ICT, SICS.
    A multicast-based distributed file system for the Internet1996In: Proceedings of the Seventh ACM SIGOPS European Workshop, 1996, 1Conference paper (Refereed)
  • 10. Hagsand, Olof
    et al.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Hanson, Kjell
    Sicsophone: A Low-delay Internet Telephony Tool2003Conference paper (Refereed)
    Abstract [en]

    The end to end delay is a critical factor in the perceived quality of service for Voice over IP applications. Sicsophone is a complete VoIP system that couples the low level features of audio hardware with a standard jitter buffer playout algorithm. Using the sound card directly eliminates intermediate buffering as well as providing fine control over timers needed by a soft real-time application such as VoIP. A statistical based approach for inserting packets into audio buffers is used in conjunction with a scheme for inhibiting unnecessary fluctuations in the system. We also present mouth-to-ear delay measurements for selected VoIP applications and show that several hundreds of milliseconds can be saved by using the techniques described in this paper. A prototype for both UNIX and Windows platforms has been implemented, demonstrating that our system adapts to network conditions whilst maintaining low delays.

  • 11. Hagsand, Olof
    et al.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Hanson, Kjell
    Sicsophone: a low-delay Internet telephony tool.2003Conference paper (Refereed)
    Abstract [en]

    The end to end delay is a critical factor in the perceived quality of service for Voice over IP applications. Sics ophone is a complete VoIP system that couples the low level features of audio hardware with a standard jitter buffer playout algorithm. Using the sound card directly eliminates intermediate buffering as well as providing fine control over timers needed by a soft real-time application such as VoIP. A statistical based approach for inserting packets into audio buffers is used in conjunction with a scheme for inhibiting unnecessary fluctuations in the system. We also present mouth-to-ear delay measurements for selected VoIP applications and show that several hundreds of milliseconds can be saved by using the techniques described in this paper. A prototype for both UNIX and Windows platforms has been implemented, demonstrating that our system adapts to network conditions whilst maintaining low delays.

  • 12.
    Hagsand, Olof
    et al.
    RISE - Research Institutes of Sweden, ICT, SICS.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Hanson, Kjell
    Sicsophone: A Low-Delay Internet Telephony Tool2002Report (Other academic)
    Abstract [en]

    The end to end delay is a critical factor in the perceived quality of service for Voice over IP applications. The described solution is a complete system-level platform and complements QoS work in the network and application areas. We describe a VoIP system that couples the low level features of audio hardware with a jitter buffer playout algorithm. Using the sound card directly eliminates intermediate buffering as well as providing fine control over timers needed by a soft real-time application such as VoIP. A statistical based approach for inserting packets into audio buffers is used in conjunction with a scheme for inhibiting unnecessary fluctuations in the system. We give comparisons for the performance of the playout algorithm against idealised playout conditions. We also present mouth to ear delay measurements for selected VoIP applications and show that several hundreds of milliseconds can be saved by using the techniques described in this paper. A prototype for both UNIX and Windows platforms (NT and 9X) has been implemented, demonstrating that our system adapts to network conditions whilst maintaining low delays.

  • 13. Hagsand, Olof
    et al.
    Más, Ignacio
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Karlsson, Gunnar
    Self-admission control for ip telephony using early quality estimation2004In: NETWORKING 2004, Networking Technologies, Services, and Protocols; Performance of Computer and Communication Networks; Mobile and Wireless Communications (Third International IFIP-TC6 Networking Conference, Athens, Greece, May 9-14, 2004. Proceedings), Springer , 2004, 1, , p. 1513p. 381-391Chapter in book (Refereed)
    Abstract [en]

    If quality of service could be provided at the transport or the application layer, then it might be deployed simply by software upgrades, instead of requiring a complete upgrade of the network infrastructure. In this paper, we propose a self-admission control scheme that does not require any network support or external monitoring schemes. We apply the admission control scheme to IP telephony as it is an important application benefiting from admission control. We predict the quality of the call by observing the packet loss over a short initial period using an in-band probing mechanism. The quality prediction is then used by the application to continue or to abort the call. Using over 9500 global IP telephony measurements, we show that it is possible to accurately predict the quality of a call. Early rejection of sessions has the advantage of saving valuable network resources plus not disturbing the on-going calls.

  • 14. Kaj, Ingemar
    et al.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Modelling the arrival process for packet audio2003In: Quality of Service in Multiservice IP Networks: Second International Workshop, QoS-IP 2003: Proceedings, 2003, 1Conference paper (Refereed)
    Abstract [en]

    Packets in an audio stream can be distorted relative to one another during the traversal of a packet switched network. This distortion can be mainly attributed to queues in routers between the source and the destination. The queues can consist of packets either from our own flow, or from other flows. The contribution of this work is a Markov model for the time delay variation of packet audio in this scenario. Our model is extensible, and show this by including sender silence suppression and packet loss into the model. By comparing the model to wide area traffic traces we show the possibility to generate an audio arrival process similar to those created by real conditions. This is done by comparing the probability density functions of our model to the real captured data.

  • 15. Leister, Wolfgang
    et al.
    Sutinen, Tiia
    Boudko, Svetlana
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Griwodz, Carsten
    Halvorsen, Pål
    An architecture for adaptive multimedia streaming to mobile nodes2008In: Proceedings of the 6th International Conference on Advances in Mobile Computing and Multimedia, 2008, 1, , p. 10Conference paper (Refereed)
    Abstract [en]

    We describe the adimus architecture which addresses the problem of maintaining the subjective quality of multimedia streaming for a mobile user. In contrast to other works, the entire end-to-end path of the video stream is considered. Adaptation mechanisms for maintaining quality include time-critical handovers, overlay routing and network estimation techniques. Our architecture is built on overlays that provides the necessary functionality for a video streaming service. The paper highlights the key components that ADIMUS advocates to support quality streaming from server to mobile client.

  • 16.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    A new proposal for congestion control in Ambient networks2005Report (Other academic)
    Abstract [en]

    This report describes the protocol design of TCP with Forward Error Correction (TCP-FEC). The performance of TCP can be significantly improved by generating and sending redundant segments in addition to the normal TCP data segments. Data losses in the network can be recovered from the redundant or the original data packets. Additionally by adding correcting codes to the TCP transmissions, both isolated and bursty losses can be handled. The advantage for the application is that the long retransmission times can be avoided if the repair can be done locally. The advantage for the network is two-fold, excessive retransmissions do not further congest the network and wireless losses can be repaired at the receiver. This technical report details TCP-FEC from a design, migration and protocol perspective in the highly heterogeneous wireless environments envisaged by the sixth framework Ambient Networks project\footnote{This work was supported by the EU Ambient Networks Project IST-2002-507134-AN}.

  • 17.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Measuring Internet telephony quality: where are we today?1999Conference paper (Refereed)
  • 18.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Quality aspects of audio communication2003Licentiate thesis, monograph (Other academic)
    Abstract [en]

    The Internet is increasingly being used to carry real-time voice traffic. Users of real-time voice services are sensitive to variable audio quality. The quality of packet audio is largely determined by the mouth-to-ear delay and the packet loss. The contribution of this thesis is to provide techniques to improve the packet audio quality: dimensioning links specifically for packet voice communication, modelling the packet audio arrival process at a receiver, measuring connectivity quality in wide area networks, and reducing delays in end systems.

  • 19.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Quality aspects of Internet telephony2009Doctoral thesis, monograph (Other academic)
    Abstract [en]

    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.

  • 20.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Real-time voice over wireless IP networks: The last challenge?2005In: INFOCOM 2005 Student Workshop, 2005, 1Conference paper (Refereed)
    Abstract [en]

    Wireless connectivity is needed to bring IP-based telephony into serious competition with the cellular infrastructure. Nevertheless, quality problems remain with wireless VoIP services, not least when used with unlicensed spectrum technologies. This submission summaries on-going work in the area of real-time voice quality issues using a combination of networking and real-user assessment techniques.

  • 21.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    VANET communication: A traffic flow approach2012Conference paper (Refereed)
  • 22.
    Marsh, Ian
    et al.
    RISE - Research Institutes of Sweden, ICT, SICS.
    Grönvall, Björn
    RISE, Swedish ICT, SICS.
    Hammer, Florian
    The design and implementation of a quality-based handover trigger2006Conference paper (Refereed)
    Abstract [en]

    Wireless connectivity is needed to bring IP-based telephony into serious competition with the existing cellular infrastructure. However it is well known that voice quality problems can occur when used with unlicensed spectrum technologies such as the popular IEEE 802.11 standards. The cellular infrastructure could provide alternative network access should users roam out of 802.11 coverage or if heavy traffic loads are encountered in the 802.11 cell. Therefore, our goal is to design a handover mechanism to switch ongoing calls to the cellular network when the 802.11 network cannot sustain sufficient call quality. We have investigated load and coverage scenarios and designed, implemented and evaluated the performance of an 802.11 quality-based trigger for the handover of voice calls to the cellular network. We show that our predictive solution addresses the coverage problem and evaluate it within a real setting.

  • 23.
    Marsh, Ian
    et al.
    RISE - Research Institutes of Sweden, ICT, SICS.
    Li, Fengyi
    Karlsson, Gunnar
    Wide Area Measurements of Voice Over IP Quality2003Report (Other academic)
    Abstract [en]

    Time, day, location and instantaneous network conditions largely dictate the quality of Voice over IP calls. In this paper we present the results of over 18000 VoIP measurements, taken from nine sites connected in a full-mesh configuration. We measure the quality of the routes on a hourly basis by transmitting a pre-recorded call between a pair of sites. We repeat the procedure for all nine sites during the one hour interval. Based on the obtained jitter, delay and loss values as defined in RFC 1889 (RTP) we conclude that the VoIP quality is acceptable for all but one of the nine sites we tested. We also conclude that VoIP quality has improved marginally since we last conducted a similar study in 1998.

  • 24.
    Marsh, Ian
    et al.
    RISE - Research Institutes of Sweden, ICT, SICS.
    Severiano, Juan Carlos Martin
    Nunes, Victor Yuri Diogo
    Maguire, Gerald Q.
    IEEE 802.11b voice quality assessment using cross-layer information2006Conference paper (Refereed)
    Abstract [en]

    This paper reports on the suitability of IEEE 802.11b networks for carrying real-time voice traffic, considering particularly the end terminals. More specifically we looked at such networks in different operating circumstances: an outdoor environment, an office environment, and the influence of competing traffic. Additionally we have investigated the link protocol in combination with the application layer. Based on over 2500 recorded sessions, it can be generally concluded that the 802.11b protocol can support real-time voice; particularly if the link transmission rate is immediately lowered after an unsuccessful initial transmission. However, we did find situations where the voice quality deteriorated below commonly accepted values, such as when competing with high-rate TCP traffic, when intervening obstacles blocked the transmission path, and with certain uses of the RTS/CTS mechanism.

  • 25.
    McNamara, Liam
    et al.
    RISE, Swedish ICT, SICS.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    CheesePi: A Raspberry Pi based measurement platform2015Conference paper (Refereed)
    Abstract [en]

    We aim to to objectively characterise the service users experience from their home Internet connections. The attributes of an Internet connection (e.g., bandwidth and loss rate) dictate the service quality that can be achieved over it. American video use is increasing rapidly with 70% of broadband users under the age of 35 getting some of their TV from online sources. Measurement of such connections is crucial, their characterisation is useful not only for human users (so people know what service they can receive) but also for the users’ devices for adaptive behaviour. Furthermore, large-scale characterisation data of individual connections can be collated into a characterisation of the whole network. In this paper we will present a distributed measurement system that we have built and the choices that comprise its design.

  • 26.
    Varela, Martin
    et al.
    RISE, Swedish ICT, SICS.
    Marsh, Ian
    RISE - Research Institutes of Sweden, ICT, SICS.
    Grönvall, Björn
    RISE, Swedish ICT, SICS.
    A Systematic Study of PESQ's Performance (from a Networking Perspective)2006Conference paper (Refereed)
    Abstract [en]

    In this paper we study, in a systematic way, how the behavior of PESQ estimations varies with the network loss process. We assess the variability of the estimations with respect to the network conditions and the speech content, and also their accuracy, by comparing the estimates with subjective assessments.

1 - 26 of 26
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