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  • 1.
    Abbas, Muhammad Tahir
    et al.
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Eklund, Johan
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Brunström, Anna
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Impact of Tunable Parameters in NB-IoT Stack onthe Energy Consumption2019In: Proceedings of Fifteenth Swedish National Computer Networking Workshop (SNCNW), 2019Conference paper (Refereed)
    Abstract [en]

    This paper studies the impact of tunable parametersin the NB-IoT stack on the energy consumption of a user equipment(UE), e.g., a wireless sensor. NB-IoT is designed to enablemassive machine-type communications for UE while providing abattery lifetime of up to 10 years. To save battery power, most oftime the UE is in dormant state and unreachable. Still, duringthe CONNECTED and IDLE state, correct tuning of criticalparameters, like Discontinuous reception (DRX), and extendedDiscontinuous reception (eDRX), respectively, are essential to savebattery power. Moreover, the DRX and eDRX actions relate tovarious parameters which are needed to be tuned in order toachieve a required UE battery lifetime. The objective of thispaper is to observe the influence of an appropriate tuning ofthese parameters to reduce the risk of an early battery drainage

  • 2.
    Ahlgren, Bengt
    et al.
    RISE SICS.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Arahamsson, Henrik
    RISE SICS.
    Brunström, Anna
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Hurtig, Per
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Latency-aware Multipath Scheduling inInformation-centric Networks2019In: Proceedings of the Fifteenth Swedish National Computer Networking Workshop (SNCNW), Luleå, Sweden. 4-5 June 2019., 2019Conference paper (Refereed)
    Abstract [en]

    We present the latency-aware multipath schedulerZQTRTT that takes advantage of the multipath opportunities ininformation-centric networking. The goal of the scheduler is touse the (single) lowest latency path for transaction-oriented flows,and use multiple paths for bulk data flows. A new estimatorcalled zero queue time ratio is used for scheduling over multiplepaths. The objective is to distribute the flow over the paths sothat the zero queue time ratio is equal on the paths, that is,so that each path is ‘pushed’ equally hard by the flow withoutcreating unwanted queueing. We make an initial evaluation usingsimulation that shows that the scheduler meets our objectives.

  • 3.
    Ahlgren, Bengt
    et al.
    RISE SICS.
    Hurtig, Per
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Abrahamsson, Henrik
    RISE SICS.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Brunström, Anna
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science. Karlstad University, Faculty of Economic Sciences, Communication and IT, Centre for HumanIT.
    Are MIRCC and Rate-based Congestion Control in ICN READY for Variable Link Capacity?2017Conference paper (Other academic)
    Abstract [en]

    Information-centric networking (ICN) has been introduced as a potential future networking architecture. ICN promises an architecture that makes information independent from lo- cation, application, storage, and transportation. Still, it is not without challenges. Notably, there are several outstanding issues regarding congestion control: Since ICN is more or less oblivious to the location of information, it opens up for a single application flow to have several sources, something which blurs the notion of transport flows, and makes it very difficult to employ traditional end-to-end congestion control schemes in these networks. Instead, ICN networks often make use of hop-by-hop congestion control schemes. How- ever, these schemes are also tainted with problems, e.g., several of the proposed ICN congestion controls assume fixed link capacities that are known beforehand. Since this seldom is the case, this paper evaluates the consequences in terms of latency, throughput, and link usage, variable link capacities have on a hop-by-hop congestion control scheme, such as the one employed by the Multipath-aware ICN Rate-based Congestion Control (MIRCC). The evaluation was carried out in the OMNeT++ simulator, and demonstrates how seemingly small variations in link capacity significantly deterio- rate both latency and throughput, and often result in inefficient network link usage. 

  • 4.
    Ahlgren, Bengt
    et al.
    RISE SICS.
    Hurtig, Per
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science. Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Abrahamsson, Henrik
    RISE SICS.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science. Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Brunström, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science. Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    ICN Congestion Control for Wireless Links2018In: IEEE WCNC 2018 Conference Proceedings / [ed] IEEE, New York: IEEE, 2018Conference paper (Refereed)
    Abstract [en]

    Information-centric networking (ICN) with its design around named-based forwarding and in-network caching holds great promises to become a key architecture for the future Internet. Still, despite its attractiveness, there are many open questions that need to be answered before wireless ICN becomes a reality, not least about its congestion control: Many of the proposed hop-by-hop congestion control schemes assume a fixed and known link capacity, something that rarely – if ever – holds true for wireless links. As a first step, this paper demonstrates that although these congestion control schemes are able to fairly well utilise the available wireless link capacity, they greatly fail to keep the link delay down. In fact, they essentially offer the same link delay as in the case with no hop-by-hop, only end- to-end, congestion control. Secondly, the paper shows that by complementing these congestion control schemes with an easy- to-implement, packet-train link estimator, we reduce the link delay to a level significantly lower than what is obtained with only end-to-end congestion control, while still being able to keep the link utilisation at a high level. 

  • 5.
    Asplund, Katarina
    et al.
    Karlstad University, Division for Information Technology.
    Brunstrom, Anna
    Karlstad University, Division for Information Technology.
    Garcia, Johan
    Karlstad University, Division for Information Technology.
    Schneyer, Sean
    Karlstad University, Division for Information Technology.
    Grinnemo, Karl-Johan
    Karlstad University, Division for Information Technology.
    PRTP: A Partially Reliable Transport Protocol for Multimedia Applications: Background, Information and Analysis1999Report (Other academic)
  • 6.
    Atxutegi, Eneko
    et al.
    University of the Basque Country, Spain.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Izurza, Andoni
    University of the Basque Country, Spain.
    Arvidsson, Åke
    Kristianstad University, Sweden.
    Liberal, Fidel
    University of the Basque Country, Spain.
    Brunström, Anna
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    On the move with TCP in current and future mobile networks2017In: Proceedings of the 2017 8th International Conference on the Network of the Future (NOF). / [ed] T. Mahmoodi, S. Secci, A. Cianfrani, F. Idzikowski, New York, USA: IEEE, 2017, p. 66-72Conference paper (Refereed)
    Abstract [en]

    Mobile wireless networks constitute an indispensable part of the global Internet, and with TCP the dominating transport protocol on the Internet, it is vital that TCP works equally well over these networks as over wired ones. This paper identifies the performance dependencies by analyzing the responsiveness of TCP NewReno and TCP CUBIC when subject to bandwidth variations related to movements in different directions. The presented evaluation complements previous studies on 4G mobile networks in two important ways: It primarily focuses on the behavior of the TCP congestion control in medium- to high-velocity mobility scenarios, and it not only considers the current 4G mobile networks, but also low latency configurations that move towards the overall potential delays in 5G networks. The paper suggests that while both CUBIC and NewReno give similar goodput in scenarios where the radio channel continuously degrades, CUBIC gives a significantly better goodput in scenarios where the radio channel quality continuously increases. This is due to CUBIC probing more aggressively for additional bandwidth. Important for the design of 5G networks, the obtained results also demonstrate that very low latencies are capable of equalizing the goodput performance of different congestion control algorithms. Only in low latency scenarios that combine both large fluctuations of available bandwidths and a mobility pattern in which the radio channel quality continuously increases can some performance differences be noticed.

  • 7.
    Atxutegi, Eneko
    et al.
    University of the Basque Country.
    Liberal, Fidel
    University of the Basque Country.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Brunström, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Centre for HumanIT. Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Arvidsson, Åke
    Kristianstad University.
    TCP Performance over Current Cellular Access: A Comprehensive Analysis2018In: Autonomous Control for a Reliable Internet of Services: Methods, Models, Approaches, Techniques, Algorithms and Tools, Springer, 2018, p. 371-400Chapter in book (Refereed)
    Abstract [en]

    Mobile internet usage has significantly raised over the last decade and it is expected to grow to almost 4 billion users by 2020. Even after the great effort dedicated to the improvement of the performance, there still exist unresolved questions and problems regarding the interaction between TCP and mobile broadband technologies such as LTE. This chapter collects the behavior of distinct TCP implementation under various network conditions in different LTE deployments including to which extent the performance of TCP is capable of adapting to the rapid variability of mobile networks under different network loads, with distinct flow types, during start-up phase and in mobile scenarios at different speeds. Loss-based algorithms tend to completely fill the queue, creating huge standing queues and inducing packet losses both under stillness and mobility circumstances. On the other side delay-based variants are capable of limiting the standing queue size and decreasing the amount of packets that are dropped in the eNodeB, but they are not able under some circumstances to reach the maximum capacity. Similarly, under mobility in which the radio conditions are more challenging for TCP, the loss-based TCP implementations offer better throughput and are able to better utilize available resources than the delay-based variants do. Finally, CUBIC under highly variable circumstances usually enters congestion avoidance phase prematurely, provoking a slower and longer start-up phase due to the use of Hybrid Slow-Start mechanism. Therefore, CUBIC is unable to efficiently utilize radio resources during shorter transmission sessions.

  • 8.
    Atxutegi, Eneko
    et al.
    University of the Basque Country.
    Liberal, Fidel
    University of the Basque Country.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Brunström, Anna
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Arvidsson, Åke
    Kristianstad University.
    Robert, Remi
    Ericsson AB.
    TCP behaviour in LTE: impact of flow start-up and mobility2016In: Wireless and Mobile Networking Conference (WMNC), 2016 9th IFIP, IEEE, 2016, p. 73-80Conference paper (Refereed)
    Abstract [en]

    Nowadays, more than two billion people uses the mobile internet, and it is expected to rise to almost 4 billion by 2020. Still, there is a gap in the understanding of how TCP and its many variants work over LTE. To this end, this paper evaluates the extent to which five common TCP variants, CUBIC, NewReno, Westwood+, Illinois, and CAIA Delay Gradient (CDG), are able to utilise available radio resources under hard conditions, such as during start-up and in mobile scenarios at different speeds. The paper suggests that CUBIC, due to its Hybrid Slow- Start mechanism, enters congestion avoidance prematurely, and thus experiences a prolonged start-up phase, and is unable to efficiently utilise radio resources during shorter transmission sessions. Still, CUBIC, Illinois and NewReno, i.e., the loss-based TCP implementations, offer better throughput, and are able to better utilise available resources during mobility than Westwood+ and CDG – the delay-based variants do. 

  • 9.
    Atxutegi, Eneko
    et al.
    University of the Basque Country, Spain.
    Liberal, Fidel
    University of the Basque Country, Spain.
    Haile, Habtegebreil Kassaye
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Brunström, Anna
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Arvidsson, Åke
    Kristianstad University, Sweden.
    On the use of TCP BBR in cellular networks2018In: IEEE Communications Magazine, ISSN 0163-6804, E-ISSN 1558-1896, no 3, p. 172-179Article in journal (Refereed)
    Abstract [en]

    TCP BBR (Bottleneck Bandwidth and Round-trip propagation time) is a new TCP variant developed at Google, and which, as of this year, is fully deployed in Googles internal WANs and used by services such as Google.com and YouTube. In contrast to other commonly used TCP variants, TCP BBR is not loss-based but model-based: It builds a model of the network path between communicating nodes in terms of bottleneck bandwidth and minimum round-trip delay and tries to operate at the point where all available bandwidth is used and the round-trip delay is at minimum. Although, TCP BBR has indeed resulted in lower latency and a more efficient usage of bandwidth in fixed networks, its performance over cellular networks is less clear. This paper studies TCP BBR in live mobile networks and through emulations, and compares its performance with TCP NewReno and TCP CUBIC, two of the most commonly used TCP variants. The results from these studies suggest that in most cases TCP BBR outperforms both TCP NewReno and TCP CUBIC, however, not so when the available bandwidth is scarce. In these cases, TCP BBR provides longer file completion times than any of the other two studied TCP variants. Moreover, competing TCP BBR flows do not share the available bandwidth in a fair way, something which, for example, shows up when shorter TCP BBR flows struggle to get its fair share from longer ones. 

  • 10.
    Baucke, Stephan
    et al.
    Ericsson Research, Germany.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Ludwig, Reiner
    Ericsson Research, Germany.
    Brunström, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Wolisz, Adam
    Department of Electrical Engineering, Technical University Berlin, Germany.
    Using Relaxed Timer Backoff to Reduce SCTP Failover TimesManuscript (Other academic)
  • 11.
    Bergman, Andreas
    et al.
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Pieskä, Marcus
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Westlinder, Simon
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Rust, Josefine
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science. Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Socket Intents Extended for SCTP: Extended Version of Socket Intents to Use the Transport Protocol SCTP2016Report (Other (popular science, discussion, etc.))
    Abstract [en]

    This report covers a project in the course Computer Engineering Project, DVAE08, at Karlstad University. The aim of the project was to modify an already existing solution for selecting the most fitting path for known traffic online, with a proactive approach instead of a reactive, called Socket Intents. The purpose of the modified version is to make the previous solution compatible with the transport protocol SCTP. This solution consists of three new implemented components; a header parser, a sniffer, and a query manager. The header parser and sniffer receive packets from the traffic and send them to one another. The query manager handles queries from the policies to the sniffer, as well as the response. Together, these components will gather information about the state of the network, and select the most fitting path that fulfill application needs. The results achieved from the modification are good work for the SCTP one-to-one type. 

  • 12.
    Bozakov, Zdravko
    et al.
    Dell EMC.
    Brunström, Anna
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Damjanovic, Dragana
    Mozilla.
    Riktor Evensen, Kristian
    Celerway.
    Fairhurst, Gorry
    University of Aberdeen, UK..
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Jones, Tom
    University of Aberdeen, UK..
    Mangiante, Simone
    Dell EMC.
    Papastergiou, Giorgos
    Simula.
    Ros, David
    Simula.
    Tüxen, Michael
    FH Münster, Germany.
    Welzl, Michael
    University of Oslo, Norway.
    Deliverable D2.1 - First Version of Low-Level Core Transport System2016Report (Refereed)
    Abstract [en]

    This document presents the first version of the low-level Core Transport System in NEAT, to be used for development of a reference implementation of the NEAT System. The design of this core transport system takes into consideration the Transport Services and the API defined in Task 1.3 and in close coordination with the overall architecture (Task 1.2). To realise the basic Transport Services provided by the API, a set of low-level transport functionalities has to be provided by the NEAT core transport system. These functionalities take the formof several building blocks, or NEAT Components, each representing an associated implementation activity. Some of the components are needed to ensure the basic operation of the NEAT System—e.g., a NEAT Flow Endpoint, a callback-based NEAT API Framework, the NEAT Logic and the functionality to Connect to a name. Some other components are needed to ensure connectivity usingMiddlebox Traversal techniques (e.g., TURN), discovery of path support for different transport protocols using Happy Eyeballs mechanisms, offering end-to end Security (e.g., (D)TLS over transport), gather statistics for the users or system administrators, and the ability to apply different policies in order to influence the decision-making process of the transport system. This document describes each of these building blocks and related design choices.

  • 13.
    Budzisz, Lukasz
    et al.
    Signal Theory and Communication Dept., Universitat Politechnìca de Catalunya.
    Ferrus, Ramon
    Signal Theory and Communication Dept., Universitat Politechnìca de Catalunya.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    An Analytical Estimation of the Failover Time in SCTP Multihoming Scenarios2007In: IEEE Wireless Communications and Networking Conference, IEEE , 2007, p. 3929-3934Conference paper (Refereed)
    Abstract [en]

    The motivation behind this paper is a need to have a more accurate estimation of the failover time in SCTP. The traditional one, commonly used in the literature, is based on the sum of the consecutive retransmission timeouts. This is not always appropriate, especially when using the SCTP multihoming feature as a basis for achieving transport layer mobility in wireless networking scenarios, where the transition time between available paths becomes a key aspect for the optimisation. Two new factors are introduced into the proposed estimation formula to reflect the influence of the network parameters and the behaviour of the most common protocol implementations. For the proposed model, we perform a best-worst case analysis, and then illustrate it with an example of a detailed estimation. Finally, we perform simulations comparing our proposal with the traditional estimation in a typical transport layer mobility scenario including long thin networks.

  • 14.
    Budzisz, Lukasz
    et al.
    Department of Signal Theory and Communications (TSC), Universitat Politecnica de Catalunya.
    Ferrús, Ramon
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Fracchia, Roberta
    Motorola Labs – Paris, France.
    Galante, Giulio
    Istituto Superiore Mario Boella.
    Casadewall, Ferran
    Department of Signal Theory and Communications (TSC), Universitat Politecnica de Catalunya.
    Towards transport-layer mobility: Evolution of SCTP multihoming2008In: Computer Communications, ISSN 0140-3664, E-ISSN 1873-703X, Vol. 31, no 5, p. 980-998Article in journal (Refereed)
    Abstract [en]

    Recently, growing availability of emerging wireless technologies has pushed the demand to integrate different wireless-network technologies such as: wireless local-area networks, cellular networks, and personal and short-range networks. The inter-working of heterogeneous radio access networks poses many technical challenges, with mobility management being one of the most important. In this paper we survey the existing proposals and show that transport-layer mobility is a viable candidate for implementing seamless handover in heterogeneous wireless access networks. Since the mobile Stream Control Transmission Protocol (mSCTP) is at the core of most relevant transport-layer mobility schemes being currently studied, we identify the key scenarios where the protocol can effectively leverage the multihoming feature to enhance handover support. Moreover, to provide the reader with a complete overview of the mSCTP's application area, we also survey the situations where the use of mSCTP-based schemes is not possible or has some limitations. Then, in one of the identified key scenarios, we investigate several challenging open issues related to path management and path-transition optimization by considering bandwidth-estimation schemes and link-layer support. Finally, we consider introducing concurrent multipath transfer (CMT) into mSCTP-based mobility schemes, as a future research direction.

  • 15.
    Cheng, Jun
    et al.
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Telco Distributed DC with Transport Protocol Enhancement for 5G Mobile Networks: A Survey2017Report (Other academic)
    Abstract [en]

    Distributed data center hosts telco virtual network functions, mixing workloads that require data transport through transport protocols with either low end-to-end latency or large bandwidth for high throughput, e.g., from tough requirements in 5G use cases. A trend is the use relatively inexpensive, off-the-shelf switches in data center networks, where the dominated transport traffic is TCP traffic. Today’s TCP protocol will not be able to meet such requirements. The transport protocol evolution is driven by transport performance (latency and throughput) and robust enhancements in data centers, which include new transport protocols and protocol extensions such as DCTCP, MPTCP and QUIC protocols and lead to intensive standardization works and contributions to 3GPP and IETF.

    By implementing ECN based congestion control instead of the packet-loss based TCP AIMD congestion control algorithm, DCTCP not only solves the latency issue in TCP congestion control caused by the switch buffer bloating but also achieves an improved performance on the packet loss and throughput. The DCTCP can also co-exist with normal TCP by applying a modern coupled queue management algorithm in the switches of DC networks, which fulfills IETF L4S architecture. MPTCP is an extension to TCP, which can be implemented in DC’s Fat tree architecture to improve transport throughput and shorten the latency by mitigating the bandwidth issue caused by TCP connection collision within the data center. The QUIC is a reliable and multiplexed transport protocol over UDP transport, which includes many of the latest transport improvements and innovation, which can be used to improve the transport performance on streaming media delivery.

    The Clos topology is a commonly used network topology in a distributed data center. In the Clos architecture, an over-provisioned fabric cannot handle full wire-speed traffic, thus there is a need to have a mechanism to handle overload situations, e.g., by scaling out the fabric. However, this will introduce more end-to-end latency in those cases the switch buffer is bloated, and will cause transport flow congestion.

    In this survey paper, DCTCP, MPTCP and QUIC are discussed as solutions for transport performance enhancement for 5G mobile networks to avoid the transport flow congestion caused by the switch buffer bloating from overloaded switch queue in data centers. 

  • 16. Christoforidis, Christos
    et al.
    Ivarsson, Niklas
    Johansson, Henrik
    Nilsson, Jonas
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science. Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    SCTPTrace: An Extension of TCPTrace for SCTP2016Report (Other (popular science, discussion, etc.))
    Abstract [en]

    When it comes to analyzing TCP data and extracting the information in such a way that it becomes viewable, there are a couple of tools that can be used. One of them is TCPTrace. TCPTrace is used to analyze special dump files created from programs such as tcpdump, snoop and WinDump. TCPTrace became published for a broader public in the late 1996 by Shawn Ostermann. Since then functionalities, changes and fixes have been implemented for example the extension to create graphs and trace UDP packets. From the dump files a trace will be done, and depending on the input from the user, TCPTrace can present this information in a number of ways such as plain text, trace files and graphs, depending on the amount of information the user is looking for. The extensive information traced will be viewed and divided for each connection found. For each connection, information such as retransmits, throughput, round trip times, bytes and packets sent and received etc. can be presented.

    This project came to be, since there has been a desire to see a tool for SCTP that provides the same functionalities as TCPTrace. The project, called SCTPTrace, aimed to implement as much of the previous TCP functionalities as possible for the SCTP protocol. 

  • 17.
    Eklund, Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Improving mSCTP-based Vertical Handovers by Increasing the Initial Congestion Window2011In: / [ed] Johan Eklund,Karl-Johan Grinnemo, Anna Brunstrom, 2011Conference paper (Refereed)
    Abstract [en]

    Wireless networks play an increasingly important role in our daily lives. However, the wireless landscape comprises serveralnetwork technologies, and which technology to use is often context dependent and varies over time: In urban areas, WiFi viaa WiFi hotspot might be the preferred access technology, while in rural areas 3G, HSPA or some other 3GPP technologymight be the only viable alternative. To this end, a large number of vertical handover solutions, ranging from link-level toapplication-level solutions, has been proposed in the past several years.We believe that the Stream Control Transmission Protocol and its extension for Dynamic Address Reconfiguration (mSCTP)make up an attractive vertical handover solution, especially in cases where there are little or no economic incentives to modify orupgrade the existing network infrastructure. Currently, we are implementing an mSCTP-based mobility management frameworkfor Android smartphones and tablets, and as part of that work we are considering ways of mitigating the effects of a handoveron ongoing transport sessions.A mobile terminal with several network interfaces may traverse different types of networks (WLAN, UMTS, HSPA).A seamless handover between these networks may be achieved by the transport protocol, Stream Control TransmissionProtocol(SCTP), with its multihoming facility (mSCTP), in conjunction with a mobility manager monitoring the availablenetwork interfaces. SCTP, a reliable transport protocol, closely related to TCP, encompasses slow-start in the beginning of asession. This slow-start phase is also activated to probe for network capacity on the target path after a handover, which couldresult in unncessarily reduced throughput in case of spare capacity in the network. The throughput degradation in could becrucial for an ongoing real time session, like video conferencing. In this study we have conducted experiments on the benefitof increasing the initial congestion window to reduce the negative impact of the slow start phase. We have by experimentsquantified the performance gain from an increasd initial window, and have seen that there a reduction in maximum transfertime for a message of about 50 % in scenarios with real-time video, compared to the default initial congestion window. Theimprovement is most prominent in sessions with high transfer delays. One easy, and thus very attractive, way to improveon the handover performance is to increase the initial congestion window on the handover target path. Although a seeminglycontroversal proposition, one should bear in mind that researchers at Google and other cloud computing companies are currentlystrongly promoting a drastic increase of TCP’s initial congestion window within IETF.

  • 18.
    Eklund, Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science. Karlstad University, Faculty of Economic Sciences, Communication and IT, Centre for HumanIT.
    Brunström, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science. Karlstad University, Faculty of Economic Sciences, Communication and IT, Centre for HumanIT.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Cheimonidis, Georgios
    Ismailov, Yuri
    Delay Penalty during SCTP Handover2011Conference paper (Refereed)
    Abstract [en]

    The rapidly growing interest in untethered Internet connections such as WLAN and 3G/4G mobile connections,calls for intelligent session management, not least in terms of handovers. As part of an effort to develop a SCTP-based session management framework, we are studying ways of improving the SCTP handover delay forreal-time traffic by optimizing the startup delay on the handover-target path. We have developed a theoretical model that predicts the transfer times of SCTP messages during the startup on a new path. This paper validates our model. It shows that the model can be used to predict message transfer times in variable bitrate flows. The paper further employs our model to study the startup delay penalty during handover for the spectrum of network conditions considered relevant for real-time traffic over mobile connections.

  • 19.
    Eklund, Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science. Karlstad University, Faculty of Economic Sciences, Communication and IT, Centre for HumanIT.
    Brunström, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science. Karlstad University, Faculty of Economic Sciences, Communication and IT, Centre for HumanIT.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Cheimonidis, Georgios
    Ismailov, Yuri
    Impact of Slow Start on SCTP Handover Performance2011In: Proceedings of 20th International Conference on Computer Communications and Networks: ICCCN2011, IEEE conference proceedings, 2011Conference paper (Refereed)
    Abstract [en]

    The rapidly growing interest in untethered Internet connections, especially in terms of WLAN and 3G/4G mobile connections, calls for intelligent session management: a mobile device should be able to provide a reasonable end-user experience despite location changes, disconnection periods and, not least, handovers. As part of an effort to develop a SCTP-based session management framework that meets these criteria, we are studying ways of improving the SCTP handover delay for real-time traffic; especially the startup delay on the connection between a mobile device and the target access point. To obtain an appreciation of the theoretically feasible gains of optimizing the startup delay on the handover-target path, we have developed a model that predicts the transfer times of SCTP messages during slow start. This paper experimentally validates our model and demonstrates that it could be used to predict the message transfer times in a variable bitrate flow by approximating the variable flow with a constant dito. It also employs our model to obtain an appreciation of the startup delay penalties incured by slow start during handovers in typical mobile, real-time traffic scenarios.

  • 20.
    Eklund, Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Grinnemo, Karl-Johan
    School of Information and Communication Technology, KTH Royal Institute of Technology.
    Baucke, Stephan
    Ericsson Research, Germany.
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Tuning SCTP Failover for Carrier Grade Telephony Signaling2010In: Computer Networks, ISSN 1389-1286, E-ISSN 1872-7069, Vol. 54, no 1, p. 133-149Article in journal (Refereed)
    Abstract [en]

    The Stream Control Transmission Protocol (SCTP) has not only been selected as the signaling transport protocol of choice in IETF SIGTRAN, the architecture that bridges circuit-switched and IP-based mobile core networks, but also plays a pivotal role in SAE/LTE, the next-generation UMTS/HSPA networks. To meet the redundancy requirements of telecom signaling traffic, SCTP includes a failover mechanism that enables rerouting of traffic from an unreachable network path to a backup path. However, the recommendations provided by IETF on how to configure the SCTP failover mechanism to meet telecom signaling requirements are kept quite general and leave much of the tuning to the telecom equipment vendor and/or operator. Several works by us and others have been carried out to study the effect of different SCTP parameters on the failover performance. The main contribution of this paper is that it gives a coherent treatment of how to configure the SCTP failover mechanism for carrier-grade telephony signaling, and provides practically usable configuration recommendations. The paper also discusses an alternate or complementary way of optimizing the SCTP failover mechanism by relaxing the exponential backoff that foregoes a retransmission timeout in SCTP. Some results showing significantly reduced failover times by use of this mechanism, with only marginal deteriorating effects on a signaling network, are discussed and analyzed in the paper.

  • 21.
    Eklund, Johan
    et al.
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science. Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Brunstrom, Anna
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Efficient Scheduling to Reduce Latency for Signaling Traffic using CMT-SCTP2016In: 27th Annual IEEE International Symposium on Personal, Indoor and Mobile Radio Communications (PIMRC), September 4-7, Valencia, Spain, IEEE Communications Society, 2016Conference paper (Refereed)
    Abstract [en]

    To mitigate delay spikes during transmission of bursty signaling traffic, concurrent multipath transmission (CMT) over several paths in parallel could be an option. Still, unordered delivery is a well known problem when concurrently transmitting data over asymmetric network paths, leading to extra delay due to Head-of-Line Blocking (HoLB). The Stream Control Transmission Protocol (SCTP), designed as a carrier for signaling traffic over IP, is currently being extended with support for CMT (CMT-SCTP). To reduce the impact of HoLB, SCTP has support for transmission of separate data flows, called SCTP streams. In this paper, we address sender scheduling to optimize latency for signaling traffic using CMT-SCTP. We present dynamic stream-aware (DS) scheduling, which utilizes the SCTP stream concept, and continuously considers the current network status as well as the data load to make scheduling decisions. We implement a DS scheduler and compare it against some existing schedulers. Our investigation suggests that DS scheduling could significantly reduce latency compared to dynamic path scheduling that does not consider streams. Moreover, we show that naive round-robin scheduling may provide low latency over symmetric network paths, but may transmit data on non-beneficial asymmetric network paths leading to increased latency. Finally, our results show that a static stream based approach, found beneficial for bulk traffic, is not appropriate for bursty signaling traffic. 

  • 22.
    Eklund, Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Implications of using a Large Initial Congestion Window to Improve mSCTP Handover Delay2012In: MOBILITY 2012 : The Second International Conference on Mobile Services, Resources, and Users / [ed] Josef Noll, University of Oslo & Movation, Norway, Alessandro Bazzi, CNR - IEIIT, Italy, IARIA , 2012, , p. 6p. 116-121Conference paper (Refereed)
    Abstract [en]

    The currently rather heterogeneous wireless landscape makes handover between different network technologies, so-called vertical handover, a key to a continued success for wireless Internet access. Recently, an extension to the Stream Control Transmission Protocol (SCTP) – the Dynamic Address Reconfiguration (DAR) extension – was standardized by IETF. This extension enables the use of SCTP for vertical handover. Still, the way vertical handover works in SCTP with DAR makes it less suitable for real-time traffic. Particularly, it takes a significant amount of time for the traffic to ramp up to full speed on the handover target path. In this paper, we study the implications of an increased initial congestion window for real-time traffic on the handover target path when competing traffic is present. The results clearly show that an increased initial congestion window could significantly reduce the transfer delay for real-time traffic, provided the fair share of the available capacity on the handover target path is sufficiently higher than the send rate required by the real-time flow. Additionally, we notice that this performance gain comes without penalizing the competing traffic.

  • 23.
    Eklund, Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Grinnemo, Karl-Johan
    Tieto Enator AB, Karlstad, Sweden.
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    On the Relation Between SACK Delay and SCTP Failover Performance for Different Traffic Distributions2008In: Proceedings on the Fifth International Conference on Broadband Communications, Networks and Systems, IEEE , 2008, p. 577-584Conference paper (Refereed)
    Abstract [en]

    The stream control transmission protocol (SCTP) is an important component in the ongoing evolution towards IP in the fixed and mobile telephone networks. It is the transport protocol being used in the ongoing deployment of IETFpsilas signaling transport (SIGTRAN) architecture for tunneling of traditional telephony signaling traffic over IP. Further SCTP represents an alternative for future SIP signaling traffic. Key to the success of SCTP is its ability to recover from network failures, in particular failed network paths. SCTP includes multihoming and a failover mechanism which should swiftly shift from a failed or unavailable network path to a backup path. However, several studies have shown that SCTPpsilas failover performance is dependent on factors both related to protocol parameters and network conditions. This paper complements these studies by providing a comprehensive evaluation of the impact of SACK delay under various traffic distributions. The results show a clear relation between the traffic distribution and the impact of the SACK delay on SCTP failover performance. Severe negative effects are observed for low intensity traffic composed of individual signaling messages. On the other hand, our results show limited impact of SACK delay for high intensity and bursty traffic. Furthermore, the results show a limited increase in network traffic by reducing the SACK delay at low traffic intensities and even less impact on network traffic at high traffic intensities. Based on these results we recommend a decrease of the SCTP SACK timer to a small value in signaling scenarios

  • 24.
    Eklund, Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    On the Use of an Increased Initial Congestion Window to Improve mSCTP Handover Performance2012In: WAINA 2012: 26th International Conference on Advanced Information Networking and Applications Workshops / [ed] Leonard Barolli, Tomoya Enokido, Fatos Xhafa, Makoto Takizawa, IEEE Press, 2012, p. 1101-1106Conference paper (Refereed)
    Abstract [en]

    With the wireless landscape being ratherheterogeneous, handover between different networktechnologies, so-called vertical handover, becomes keyto a continued success for wireless Internet access.Recently, an extension to the Stream Control Trans-mission Protocol (SCTP) – the Dynamic Address Re-configuration (DAR) extension – was standardized bythe IETF. This extension enables the use of SCTPfor vertical handover. Still, the way vertical handoverworks in SCTP with DAR makes it less suitable for real-time traffic. Particularly, it takes a significant amountof time for the traffic to ramp up to full speed on thehandover target path. In this paper, we study the ex-tent to which an increased initial congestion window onthe handover target path decreases the transfer delayspikes in real-time video traffic experienced during avertical handover. The impact on both standard andhigh-definition video traffic is considered. The resultsof our study suggest that an increased initial congestionwindow does indeed significantly decrease the spikes inthe video traffic. However, the results also indicate thatit does not resolve the problem altogether.

  • 25.
    Eklund, Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Grinnemo, Karl-Johan
    School of Information and Communication Technology, KTH Royal Institute of Technology.
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Theoretical Analysis of an Ideal Startup Scheme in Multihomed SCTP2010In: Networked Services and Applications - Engineering, Control and Management: 16th EUNICE/IFIP WG 6.6 Workshop, EUNICE 2010, Trondheim, Norway, June 28-30, 2010 / [ed] Finn Arve Aagesen, Svein Johan Knapskog, Springer , 2010, p. 155-166Conference paper (Refereed)
    Abstract [en]

    SCTP congestion control includes the slow-start mechanism to probe the network for available bandwidth. In case of a path switch in a multihomed association, this mechanism may cause a sudden drop in throughput and increased message delays. By estimating the available bandwidth on the alternate path it is possible to utilize a more efficient startup scheme. In this paper, we analytically compare and quantify the degrading impact of slow start in relation to an ideal startup scheme. We consider three different scenarios where a path switch could occur. Further, we identify relevant traffic for these scenarios. Our results point out that the most prominent performance gain is seen for applications generating high traffic loads, like video conferencing. For this traffic, we have seen reductions in transfer time of more than 75% by an ideal startup scheme. Moreover, the results show an increasing impact of an improved startup mechanism with increasing RTTs

  • 26.
    Eklund, Johan
    et al.
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Brunström, Anna
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    On the Impact of Data Scheduling to Reduce Latency for Telephony Signaling Traffic using CMT-SCTP2015In: The 11th Swedish National Computer Networking Workshop (SNCNW), Karlstad, Sweden, May 28-29, 2015., 2015Conference paper (Other academic)
    Abstract [en]

    SCTP is a transport protocol targeted for telephony signaling traffic. Although SCTP from its inception supported multihoming, it has until now not supported concurrent mul- tipath transfer. However, this is about to change: Currently a standard for concurrent multipath transfer is underway. Since it is unclear whether concurrent multipath transfer could reduce transmission latency, this paper evaluates two algorithms for scheduling signaling traffic for transmission. We find that the mechanisms may provide good performance and to some extent enables for service differentiation. Still, the results indicate that unpredictable traffic require dynamic scheduling.

  • 27.
    Eklund, Johan
    et al.
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Brunström, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Stream-aware Scheduling to Improve Latency for Signaling Traffic using CMT-SCTP2016Conference paper (Refereed)
    Abstract [en]

    To mitigate delay spikes during transmission ofbursty signaling traffic, concurrent multipath transmission(CMT) over several paths in parallel could be an option. Still,unordered delivery is a well known problem when transmittingover asymmetric network paths, leading to extra delay due toHead-of-Line Blocking (HoLB). The Stream Control TransmissionProtocol (SCTP) is designed as a carrier for signaling trafficover IP to reduce the impact of HoLB. SCTP has support fortransmission of separate flows, called SCTP streams. SCTP iscurrently being extended with support for CMT (CMT-SCTP). Inthis paper, we address sender scheduling to optimize latency forsignaling traffic over CMT-SCTP. We present dynamic streamaware(DS) scheduling, which utilizes the SCTP stream concept,and continuously considers the current network status as well asthe data load to make scheduling decisions. We implement a DSScheduler and compare it against a dynamic path (DP) schedulerthat does not consider streams. Our investigation shows that DSscheduling could significantly reduce latency compared to a DPscheduler.

  • 28.
    Eklund, Johan
    et al.
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013). Karlstad University, Faculty of Economic Sciences, Communication and IT, Centre for HumanIT.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Brunström, Anna
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Using multiple paths in SCTP to reduce latency for signaling traffic2018In: Computer Communications, ISSN 0140-3664, E-ISSN 1873-703X, Vol. 129, p. 184-196Article in journal (Refereed)
    Abstract [en]

    The increase in traffic volumes as well as the heterogeneity in network infrastructure in the upcoming 5G cellular networks will lead to a dramatic increase in volumes of control traffic, i.e., signaling traffic, in the networks. Moreover, the increasing number of low-power devices with an on-off behavior to save energy will generate extra control traffic. These increased traffic volumes for signaling traffic, often generated as bursts of messages, will challenge the signaling application timing requirements on transmission. One of the major transport protocols deployed for signaling traffic in cellular networks is the Stream Control Transmission Protocol (SCTP), with support for multiple paths as well as for independent data flows. This paper evaluates transmission over several paths in SCTP to keep the latency low despite increasing traffic volumes. We explore different transmission strategies and find that concurrent multipath transfer over several paths will significantly reduce latency for transmission over network paths with the same or similar delay. Still, over heterogeneous paths, careful, continuous sender scheduling is crucial to keep latency low. To this end, we design and evaluate a sender scheduler that considers path characteristics as well as queuing status and data flows of different priority to make scheduling decisions. Our results indicate that by careful dynamic sender scheduling, concurrent multipath transfer could lead to reduced latency for signaling traffic irrespective of path or traffic characteristics.

  • 29.
    Fairhurst, Gorry
    et al.
    University of Aberdeen, UK..
    Jones, Tom
    University of Aberdeen, UK..
    Bozakov, Zdravko
    Dell EMC, Ireland.
    Brunström, Anna
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Damjanovic, Dragana
    Mozilla.
    Eckert, Toerless
    CISCO.
    Evensen, Kristian Riktor
    Celerway.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Fosselie Hansen, Audun
    Celerway.
    Khademi, Naeem
    University of Oslo, Norway.
    Mangiante, Simon
    Dell EMC.
    McManus, Patrick
    Mozilla.
    Papastergiou, Giorgos
    Simula.
    Ros, David
    Simula.
    Tüxen, Michael
    FH Münster, Germany.
    Vyncke, Eric
    CISCO.
    Welzl, Michael
    University of Oslo, Norway.
    Deliverable D1.1 - NEAT Architecture2016Report (Refereed)
    Abstract [en]

    Ossification of the Internet transport-layer architecture is a significant barrier to innovation of the Internet. Such innovation is desirable for many reasons. Current applications often need to implement their own mechanisms to receive the transport service they need, but many do not have the breadth of adapting to all possible network characteristics. An updated transport architecture can do much to make the Internet more flexible and extensible. New ground-breaking services often require different or updated transport protocols, could benefit from better signalling between application and network, or desire a more flexible choice of which network path is used for which traffic. This document therefore proposes a new transport architecture. Such architecture lowers the barrier to service innovation by proposing a “transport system”, the NEAT System, that can leverage the rich set of available transport protocols. It paves the way for an architectural change of the Internet where new transport-layer services can seamlessly be integrated and quickly made available, minimising deployment difficulties, and allowing Internet innovators to take advantage of them wherever possible. The document provides a survey of the state-of-the-art to identify the architectural obstacles to, and opportunities for, evolution of the transport layer. It also details a set of general requirements for a new transport architecture. This new architecture is motivated by a set of use-cases, followed by a description of the NEAT architecture for a transport system, designed to permit applications to select appropriate transports based on their needs and the available transport services.

  • 30.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Economic Sciences, Communication and IT.
    A Study of Partially Reliable Transport Protocols for Soft Real-Time Applications2002Report (Other academic)
    Abstract [en]

    The profileration of multimedia applications, such as streaming video, teleconferencing,and interactive gaming has created a tremendous challenge for the traditional transportprotocols of the Internet – UDP and TCP. Specifically, many multimedia applicationsare examples of soft real-time applications. They have often relatively stringent requirementsin terms of delay and delay jitter, but typically tolerate a limited packet loss rate.In recognition of the transport service requirements of soft real-time applications,this thesis studies the feasibility of using retransmission based, partially reliable transportprotocols for these applications. The thesis studies ways of designing retransmissionbased, partially reliable transport protocols that are congestion aware and TCP compatible.Furthermore, the transport protocols should provide a service that, in terms ofperformance metrics such as throughput, delay, and delay jitter, are suitable for soft realtimeapplications. The thesis work comprises the design, analysis, and evaluation oftwo retransmission based, partially reliable transport protocols: PRTP and PRTP-ECN.Extensive simulations have been carried out on PRTP as well as PRTP-ECN. These simulationshave in part been complemented by some theoretical analysis. The results ofthe simulations and the analysis suggest that substantial reductions in delay jitter andimprovements in throughput can indeed be obtained with both PRTP and PRTP-ECN ascompared to TCP. While PRTP reacted too slowly to congestion to be TCP-friendly andaltogether fair, PRTP-ECN was found to be both TCP-friendly and reasonably fair.The thesis work also comprises an extensive survey on retransmission based, partiallyreliable transport protocols. Based on this survey, we have proposed a taxonomyfor these protocols. The taxonomy considers two dimensions of retransmission based,partially reliable transport protocols: the transport service, and the error control scheme.

  • 31.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    A Study of Partially Reliable Transport Protocols for Soft Real-Time Applications2002Licentiate thesis, comprehensive summary (Other academic)
    Abstract [en]

    The profileration of multimedia applications, such as streaming video, teleconferencing, and interactive gaming has created a tremendous challenge for the traditional transport protocols of the Internet – UDP and TCP. Specifically, many multimedia applications are examples of soft real-time applications. They have often relatively stringent require- ments in terms of delay and delay jitter, but typically tolerate a limited packet loss rate.

    In recognition of the transport service requirements of soft real-time applications, this thesis studies the feasibility of using retransmission based, partially reliable trans- port protocols for these applications. The thesis studies ways of designing retransmis- sion based, partially reliable transport protocols that are congestion aware and TCP com- patible. Furthermore, the transport protocols should provide a service that, in terms of performance metrics such as throughput, delay, and delay jitter, are suitable for soft real- time applications. The thesis work comprises the design, analysis, and evaluation of two retransmission based, partially reliable transport protocols: PRTP and PRTP-ECN. Extensive simulations have been carried out on PRTP as well as PRTP-ECN. These sim- ulations have in part been complemented by some theoretical analysis. The results of the simulations and the analysis suggest that substantial reductions in delay jitter and improvements in throughput can indeed be obtained with both PRTP and PRTP-ECN as compared to TCP. While PRTP reacted too slowly to congestion to be TCP-friendly and altogether fair, PRTP-ECN was found to be both TCP-friendly and reasonably fair.

    The thesis work also comprises an extensive survey on retransmission based, par- tially reliable transport protocols. Based on this survey, we have proposed a taxonomy for these protocols. The taxonomy considers two dimensions of retransmission based, partially reliable transport protocols: the transport service, and the error control scheme. 

  • 32.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Economic Sciences, Communication and IT.
    Taxonomy and Survey of Retransmission Based Partially Reliable Transport Protocols2002Report (Other academic)
    Abstract [en]

    The mismatch between the services offered by the two standard transport protocols inthe Internet, TCP and UDP, and the services required by distributed multimedia applicationshas led to the development of a large number of partially reliable transportprotocols. That is, protocols which in terms of reliability places themselves betweenTCP and UDP. This paper presents a taxonomy for retransmission based, partially reliabletransport protocols, i.e., the subclass of partially reliable transport protocols thatperforms error recovery through retransmissions. The taxonomy comprises two classificationschemes: one that classifies retransmission based, partially reliable transportprotocols with respect to the reliability service they offer and one that classifies themwith respect to their error control scheme. The objective of our taxonomy is fourfold:to introduce a unified terminology; to provide a framework in which retransmissionbased, partially reliable transport protocols can be examined, compared, and contrasted;to make explicit the error control schemes used by these protocols; and, finally, to gainnew insights into these protocols and thereby suggest avenues for future research. Basedon our taxonomy, a survey was made of existing retransmission based, partially reliabletransport protocols. The survey shows how protocols are categorized according to ourtaxonomy, and exemplifies the majority of reliability services and error control schemesdetailed in our taxonomy.

  • 33.
    Grinnemo, Karl-Johan
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Transport Services for Converged IP Networks Protocols for Soft Real-Time Applications in IP Networks2010Book (Refereed)
  • 34.
    Grinnemo, Karl-Johan
    et al.
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Bozakov, Zdravko
    Dell EMC.
    Brunström, Anna
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Isabel Bueno, María
    Damjanovic, Dragana
    Mozilla.
    Rikter Evensen, Kristian
    Celerway.
    Fairhurst, Gorry
    University of Aberdeen, UK..
    Hansen, Audun
    Celerway.
    Hayes, David
    University of Oslo, Norway.
    Hurtig, Per
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Khademi, Naeem
    University of Oslo, Norway.
    Mangiante, Simone
    Dell EMC.
    Mohideen, Althaff
    University of Aberdeen, UK..
    Rajiullah, Mohammad
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Ros, David
    Simula.
    Rüngeler, Irene
    FH Münster, Germany.
    Santos, Ricardo
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Secchi, Raffaello
    University of Aberdeen, UK..
    Christian Tangenes, Tor
    Tüxen, Micheal
    FH Münster, Germany.
    Weinrank, Felix
    FH Münster, Germany.
    Welzl, Michael
    University of Oslo, Norway.
    Deliverable D3.1 - Initial Report on the Extended Transport System2017Report (Refereed)
    Abstract [en]

    The NEAT System offers an enhanced API for applications that disentangles them from the actual transport protocol being used. The system also enables applications to communicate their service requirements to the transport system in a generic, transport-protocol independent way. Moreover, the architecture of the NEAT System promotes the evolution of new transport services. Work Package 3 (WP3) enhances and extends the core parts of the NEAT Transport. Efforts have been devoted to developing transport-protocol mechanisms that enable a wider spectrum of NEAT Transport Services, and that assist the NEAT System in facilitating several of the commercial use cases. Work has also started on the development of optimal transport-selection mechanisms; mechanisms that enable for the NEAT System to make optimal transport selections on the basis of application requirements and network measurements. Lastly, another research activity has been initiated on how to use SDN to signal application requirements to routers, switches, and similar network elements. This document provides an initial report on all these WP3 activities—both on completed and on near-termplanned work.

  • 35.
    Grinnemo, Karl-Johan
    et al.
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Bozakov, Zdravko
    Dell EMC.
    Brunström, Anna
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Isabel Sanchez Bueno, Maria
    Dreibholz, Thomas
    Simula.
    Rikter Evensen, Kristian
    Celerway.
    Fairhurst, Gorry
    University of Aberdeen, UK..
    Fosselie Hansen, Audun
    Celerway.
    Hayes, David
    University of Oslo, Norway.
    Hurtig, Per
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Rajiullah, Mohammad
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science (from 2013).
    Jones, Tom
    University of Aberdeen, UK..
    Ros, David
    Simula.
    Rozensztrauch, Tomasz
    Celerway.
    Tüxen, Michael
    FH Münster, Germany.
    Vyncke, Eric
    CISCO.
    Deliverable D3.3 - Extended Transport System and Transparent Support of Non-NEAT Applications2017Report (Refereed)
    Abstract [en]

    This deliverable summarises and concludes our work in Work Package 3 (WP3) to extend the transport services provided by the NEAT System developed in Work Package 2, and to enable non-NEAT applications to harness the transport services offered by NEAT. We have demonstrated how a policy- and information-based selection of transport protocol by NEAT could provide a more efficient transport service for web applications. The information on which NEAT makes its transport selection decisions resides in the Characteristics Information Base (CIB). The CIB is populated by various CIB sources, and in WP3 we have designed, implemented, and evaluated various CIB sources, including meta data from mobile broadband networks, passive measurements, IPv6 Provisioning Domain protocols and the Happy Eyeballs mechanism, which caches the outcome of its connection attempts. A key property of NEAT is that it not only “vertically” decouples applications from transport protocols, but also “horizontally”. Particularly, it enables applications to harness information about resource availability and policies from Software Defined Networking (SDN) controllers in managed networks, without these applications actually being SDN-aware. To extend the use of NEAT to non-NEAT applications, we have implemented a BSDcompatible sockets API on top of NEAT and a NEAT proxy that intercepts and replaces standard TCP connections with NEAT flows, i.e., with the transport solutions deemed most appropriate by NEAT.We have also proposed a way for non-NEAT applications to make use of NEAT through the deployment of NEAT-enabled virtual appliances in SDN-controlled networks: connections from these applications are routed via an SDN-controlled proxy that terminates the original connection and replaces it with a NEAT-selected connection.

  • 36.
    Grinnemo, Karl-Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    A Simulation Based Performance Analysis of a TCP Extension for Best Effort Multimedia Applications2002In: Proceedings: 35th Annual Simulation Symposium, IEEE , 2002, p. 327-336Conference paper (Refereed)
    Abstract [en]

    Since TCP is considered unsuitable for the majority of the emerging multimedia applications, these applications primarily use UDP transport together with proprietary congestion control schemes that have better jitter and throughput characteristics. A common problem with these congestion control schemes is that they often exhibit a TCP-unfriendly and unfair behavior. As the number of applications that uses this kind of schemes increases, this could become a serious threat to the stability and performance of the Internet. In an attempt to make TCP a viable alternative to some best-effort multimedia applications, we have proposed an extension to TCP - PRTP-ECN. The performance of PRTP-ECN has been compared with TCP in an extensive factorial simulation experiment. This paper gives a detailed description of this simulation experiment with an emphasis on its statistical design and analysis. The analysis of the experiment includes, among other things, a series of ANOVA tests. These tests indicate that PRTP-ECN gives significant reductions in average interarrival jitter, while at the same time leads to improvements in average throughput and goodput, and better link utilization. In addition, the analysis suggests that PRTP-ECN is almost as fair as TCP and exhibits a TCP-friendly behavior

  • 37.
    Grinnemo, Karl-Johan
    et al.
    Karlstad University, Division for Information Technology.
    Brunstrom, Anna
    Karlstad University, Division for Information Technology.
    A Survey of TCP-Friendly Congestion Control Mechanisms for Multimedia Traffic2003Report (Other academic)
    Abstract [en]

    The stability and performance of the Internet to date have in a large part been due tothe congestion control mechanism employed by TCP. However, while the TCP congestioncontrol is appropriate for traditional applications such as bulk data transfer, it hasbeen found less than ideal for multimedia applications. In particular, audio and videostreaming applications have difficulties managing the rate halving performed by TCP inresponse to congestion. To this end, the majority of multimedia applications use eithera congestion control scheme which reacts less drastic to congestion and therefore oftenis more aggressive than TCP, or, worse yet, no congestion control whatsoever. Sincethe Internet community strongly fears that a rapid deployment of multimedia applicationswhich do not behave in a fair and TCP-friendly manner could endanger the currentstability and performance of the Internet, a broad spectrum of TCP-friendly congestioncontrol schemes have been proposed. In this report, a survey over contemporary proposalsof TCP-friendly congestion control mechanisms for multimedia traffic in the Internetis presented. A classification scheme is outlined which shows how the majority of theproposed congestion control schemes emanate from a relatively small number of designprinciples. Furthermore, we illustrate how these design principles have been applied ina selection of congestion control scheme proposals and actual transport protocols.

  • 38.
    Grinnemo, Karl-Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Enhancing TCP for Applications with Soft Real-Time Constraints2001In: Multimedia systems and applications IV: 21-22 August 2001, Denver, USA / [ed] Andrew G. Tescher, Bhaskaran Vasudev, V. Michael Bove, Jr., Bellingham: SPIE , 2001, p. 18-31Conference paper (Refereed)
    Abstract [en]

    Experiences in the use of the Internet as a delivery medium for multimedia-based applications have revealed serious deficiencies in its ability to provide the QoS of Multimedia Applications. We propose an extension to TCP that addresses the QoS requirements of applications with soft real-time constraints. Although, TCP has been found unsuitable for real-time applications, it can with minor modifications be adjusted to better comply with the QoS needs of applications with soft real-time requirements. Enhancing TCP with support for this group of applications is important since the congestion control mechanism of TCP assures stability of the Internet. In contrast, specialized multimedia protocols that lack appropriate congestion control can never be deployed on a large scale basis. Two factors of great importance for applications with soft real time constraints are jitter and throughput. By relaxing the reliability offered by TCP, the extension gives better jitter characteristics and an improved throughput. The extension only needs to be implemented at the receiving side. The reliability provided is controlled by the receiving application, thereby allowing a flexible tradeoff between different QoS parameters. In this paper, our TCP extension is presented and analyzed. The analysis investigates how the different application-controlled parameters influence performance. Our analysis is supported by a simulation study that investigates the tradeoff between interarrival jitter, throughput, and reliability. The simulation results also confirm that the extended version of TCP still behaves in a TCP-friendly manner

  • 39.
    Grinnemo, Karl-Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Evaluation of the QoS Offered by PRTP-ECN: A TCP Compliant Partially Reliable Transport Protocol2001In: Proceedings: IWQoS '01 Proceedings of the 9th International Workshop on Quality of Service, London: Springer , 2001, p. 217-234Conference paper (Refereed)
    Abstract [en]

    The introduction of multimedia in the Internet imposes new QoS requirements on existing transport protocols. Since neither TCP nor UDP comply with these requirements, a common approach today is to use RTP/UDP and to relegate the QoS responsibility to the application. Even though this approach has many advantages, it also entails leaving the responsibility for congestion control to the application. Considering the importance of efficient and reliable congestion control for maintaining stability in the Internet, this approach may prove dangerous. Improved support at the transport layer is therefore needed. In this paper, a partially reliable transport protocol, PRTP-ECN, is presented. PRTP-ECN is a protocol designed to be both TCP-friendly and to better comply with the QoS requirements of applications with soft real-time constraints. This is achieved by trading reliability for better jitter characteristics and improved throughput. A simulation study of PRTP-ECN has been conducted. The outcome of this evaluation suggests that PRTPECN can give applications that tolerate a limited amount of packet loss significant reductions in interarrival jitter and improvements in throughput as compared to TCP. The simulations also verified the TCP-friendly behavior of PRTP-ECN

  • 40.
    Grinnemo, Karl-Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Impact of Traffic Load on SCTP Failovers in SIGTRAN2005Conference paper (Other academic)
    Abstract [en]

    With Voice over IP (VoIP) emerging as a viable alternative to the traditional circuit-switched telephony, it is vital that the two are able to intercommunicate. To this end, the IETF Signaling Transport (SIGTRAN) group has defined an architecture for seamless transportation of SS7 signaling traffic between a VoIP network and a traditional telecom network. However, at present, it is unclear if the SIGTRAN architecture will, in reality, meet the SS7 requirements, especially the stringent availability requirements. The SCTP transport protocol is one of the core components of the SIGTRAN architecture, and its failover mechanism is one of the most important availability mechanisms of SIGTRAN. This paper studies the impact of traffic load on the SCTP failover performance in an M3UA-based SIGTRAN network. The paper shows that cross traffic, especially bursty cross traffic such as SS7 signaling traffic, could indeed significantly deteriorate the SCTP failover performance. Furthermore the paper stresses the importance of configuring routers in a SIGTRAN network with relatively small queues. For example, in tests with bursty cross traffic, and with router queues twice the bandwidth-delay product, failover times were measured which were more than 50% longer than what was measured with no cross traffic at all. Furthermore, the paper also identifies some properties of the SCTP failover mechanism that could, in some cases, significantly degrade its performance

  • 41.
    Grinnemo, Karl-Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    On the Use of CMT-SCTP to Improve the Startup Behavior of PSTN Signaling Traffic2013Conference paper (Other academic)
    Abstract [en]

    There is currently work going on at IETF to standardize concurrent multipath transfer, i.e., simultaneous transfer of data over several network paths, for SCTP. This paper studies whether or not SCTP extended with concurrent multipath transfer (CMT-SCTP) could provide a faster startup behavior than standard SCTP. The paper complements previous work on CMT-SCTP, and extends it to PSTN signaling traffic. The paper suggests that CMT-SCTP could give a faster startup behavior over symmetrical paths, i.e., paths with similar bandwidths and round-trip times, but that its behavior is sensitive to differences in round-trip time between the paths. The paper also indicates that signaling traffic properties such as message size and burstiness contribute marginally to the startup behavior of CMT-SCTP. 

  • 42.
    Grinnemo, Karl-Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science. Karlstad University, Faculty of Health, Science and Technology (starting 2013).
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    On the Use of MPTCP to Reduce Latency for Cloud Applications2014Conference paper (Other academic)
    Abstract [en]

    As mobile cloud usage has become more and more prevalent, with more people depending on the cloud for both their work and their leisure, cloud-to-end-user latency have risen to become a key issue. One oft-forgotten way to cost effectively improve the quality of experience is to simultaneously make use of the several network interfaces, e.g., WiFi and 3G/4G interfaces, available on most of todays mobile devices including smartphones and tablets; that is, to employ multihoming. This paper provides an initial evaluation of the latency characteristics of the multipath extensions to TCP, Multipath TCP (MPTCP), that are currently being standardized by IETF. In particular, the paper considers the possible reductions in latency that could be obtained by using MPTCP and multiple network paths between a cloud service and a mobile end user. Traffic from three cloud applications are studied, Netflix, Google Maps, and Google Docs, representing typical high-, mid-, and low- intensity cloud-to-end-user traffic. The result suggests that significant latency reductions are indeed possible, however, primarily for high- intensity video traffic such as Netflix. 

  • 43.
    Grinnemo, Karl-Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Performance of SCTP-controlled Failovers in M3UA-based SIGTRAN Networks2004In: Simulations Series, La Jolla, Ca: Society for Computer Simulation , 2004, p. 86-91Conference paper (Refereed)
    Abstract [en]

    There are some large economic, operational, and, to some extent, technicalincentives to replace the traditional telecom network with IP. However,such a large transition will not happen overnight maybe never. Meanwhile,IP-based and traditional TDM-based telephony will have to co-exist. To addressthis situation, the IETF SIGTRAN working group has developed an architecturefor transportation of Signaling System No. 7 (SS7) traffic over IP. Still, itremains to be shown that the introduction of the SIGTRAN architecture will notsignificantly deteriorate the performance of SS7. To this end, this paperevaluates the failover performance in SIGTRAN networks. Specifically, the paperevaluates the performance of SCTP-controlled failovers in M3UA-based SIGTRAN networks.The paper suggests that in order to obtain a failover performance with SCTPcomparable to that obtained in traditional TDM-based SS7 systems, SCTP has to abandonmany of the configuration recommendations of RFC 2960 and become much more aggressivein its failover behavior. Furthermore, the paper suggests that the SCTP parameterPath.Max.Retrans has a major impact on the SCTP failover performance. Our evaluationalso indicates that for those path propagation delays envisioned in future SIGTRAN networks,the impact of the path propagation delay on the failover performance is marginal

  • 44.
    Grinnemo, Karl-Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Some Observations on the Performance of SCTP-controlled Failovers in M3UA-based SIGTRAN Networks2004In: Proceedings of the Second Swedish National Computer Networking Workshop: SNCNW2004, Karlstad, 2004Conference paper (Refereed)
    Abstract [en]

    With Voice over IP (VoIP) emerging as a viable alternative totraditional circuit-switched telephony, it is vital that the twoare able to intercommunicate. To this end, the IETF SignalingTransport (SIGTRAN) group has devised an architecture forseamless transportation of SS7 signaling traffic between aVoIP network and a traditional telecom network. However,at present, it is unclear if the SIGTRAN architecture will,in reality, meet the SS7 requirements, especially the stringentavailability requirements.The SCTP transport protocol is one of the core componentsof the SIGTRAN architecture, and its failover mechanism isone of the most important of the availability mechanisms ofSIGTRAN. Currently, we are studying the performance ofSCTP-controlled failovers, and although this study is not yetcompleted some observations have been made on propertiesof the SCTP failover mechanism which could impede on itsperformance. This paper reports on these observations, andexplains their causes.

  • 45.
    Grinnemo, Karl-Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT.
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT.
    Towards the Next Generation Network: The Softswitch Solution2006Report (Other academic)
    Abstract [en]

    Over the course of the last fifteen years, the telecommunication market has undergone dramatic changes. In the beginning of the nineties, the market essentially comprised a number of national monopolies. Today, yesterday's monopolies are under siege, and the incumbent operators face strong competition from newly established operators. Furthermore, in recent years broadband-based VoIP providers have entered the telecommunication market as worthy contenders to traditional operators. To be able to survive and thrive in this new, much more competitive, market, traditional wireless and wireline operators have to reduce their capital and operational expenditures. They also need to provide new revenue-generating applications and services. To this end, a large number

    of traditional operators has replaced, or seriously consider to replace, their legacy circuit-switched fixed and cellular core networks with IP. As a first step in the migration from circuit-switched technologies to IP, the softswitch solution has evolved. This report provides a comprehensive treatment of the softswitch solution from a technical viewpoint. Additionally, the report

    concludes with a brief discussion of the migration steps following the softswitch solution. In particular, an overview of the 3GPP IP Multimedia Subsystem (IMS) is given.

  • 46.
    Grinnemo, Karl-Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Brunstrom, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    Andersson, Torbjörn
    Tieto Enator AB, Karlstad.
    Performance Benefits of Avoiding Head-of-Line Blocking in SCTP2005In: Autonomic and Autonomous Systems and International Conference on Networking and Services, 2005. ICAS-ICNS 2005. Joint International Conference on, IEEE , 2005, p. 44-Conference paper (Refereed)
    Abstract [en]

    Mitigating the effects of head-of-line blocking (HoLB) was one of the major reasons the IETF SIGTRAN working group developed SCTP, a new transport protocol for PSTN signaling traffic, in the first place. However, studies of the impact of HoLB blocking on TCP and SCTP have given ambiguous results as to whether HoLB has, in fact, any significantly deteriorating effect on transmission delay. To this end, we have carried out a detailed experimental study on the quantitative effects of HoLB. Our study suggests that although HoLB could indeed incur a substantial delay penalty on a small fraction of the messages in an SCTP session, it has only a marginal impact on the average end-to-end transmission delay. We only observed improvements in the range of O% to 18% in average message transmission delay of using unordered delivery as compared to ordered delivery. Furthermore, there was a large variability in between different test runs, which often made the impact of HoLB statistically insignificant

  • 47.
    Grinnemo, Karl-Johan
    et al.
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Brunstrom, Anna
    Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Cheng, Jun
    Ericsson AB, Stockholm, Sweden.
    Using Concurrent Multipath Transfer to Improve the SCTP Startup Behavior for PSTN Signaling Traffic2014In: 2014 28TH INTERNATIONAL CONFERENCE ON ADVANCED INFORMATION NETWORKING AND APPLICATIONS WORKSHOPS (WAINA) / [ed] Barolli, L; Li, KF; Enokido, T; Xhafa, F; Takizawa, M, IEEE Press, 2014, p. 772-778Conference paper (Refereed)
    Abstract [en]

    Although latency in the Internet has gained much attention in the research community, the latency issues of mobile control signaling have received less attention, and this all the while many telecom operators are experiencing a several-hundred percent increase in signaling traffic over only a couple of years. We believe one way to address both the latency and increased signaling load of mobile networks, is to exploit concurrent transfer of signaling traffic over several paths a.k.a. concurrent multipath transfer. This paper studies whether or not SCTP extended with concurrent multipath transfer (CMT-SCTP) could provide a faster startup behavior than standard SCTP. The paper complements previous work on CMT-SCTP, and extends it to PSTN signaling traffic. The paper suggests that CMT-SCTP could give a faster startup behavior over paths with similar bandwidths and round-trip times, but that its behavior is sensitive to differences in round-trip time between the paths. Moreover, the paper suggests that provided CMT-SCTP is configured with large enough send and receive buffers, it could provide a faster startup behavior than standard SCTP over a multipath association, in spite of some of the paths having a packet-loss rate of several percent.

  • 48.
    Grinnemo, Karl-Johan
    et al.
    Karlstad University, Division for Information Technology.
    Brunstrom, Anna
    Karlstad University, Division for Information Technology.
    Garcia, Johan
    Karlstad University, Division for Information Technology.
    A Taxonomy and Survey of Retransmission Based Partially Reliable Transport Protocols2002Report (Other academic)
    Abstract [en]

    The mismatch between the services offered by the two standard transport protocols inthe Internet, TCP and UDP, and the services required by distributed multimedia applicationshas led to the development of a large number of partially reliable transportprotocols. That is, protocols which in terms of reliability places themselves betweenTCP and UDP. This paper presents a taxonomy for retransmission based, partially reliabletransport protocols, i.e., the subclass of partially reliable transport protocols thatperforms error recovery through retransmissions. The taxonomy comprises two classificationschemes: one that classifies retransmission based, partially reliable transportprotocols with respect to the reliability service they offer and one that classifies themwith respect to their error control scheme. The objective of our taxonomy is fourfold:to introduce a unified terminology; to provide a framework in which retransmissionbased, partially reliable transport protocols can be examined, compared, and contrasted;to make explicit the error control schemes used by these protocols; and, finally, to gainnew insights into these protocols and thereby suggest avenues for future research. Basedon our taxonomy, a survey was made of existing retransmission based, partially reliabletransport protocols. The survey shows how protocols are categorized according to ourtaxonomy, and exemplifies the majority of reliability services and error control schemesdetailed in our taxonomy.

  • 49.
    Grinnemo, Karl-Johan
    et al.
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science. Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    Brunström, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science. Karlstad University, Faculty of Health, Science and Technology (starting 2013), Department of Mathematics and Computer Science.
    A First Study on Using MPTCP to Reduce Latency for Cloud Based Mobile Applications2015In: 2015 IEEE SYMPOSIUM ON COMPUTERS AND COMMUNICATION (ISCC), IEEE, 2015, p. 64-69Conference paper (Refereed)
    Abstract [en]

    Currently, Multipath TCP (MPTCP) – a modifica- tion to standard TCP that enables the concurrent use of several network paths in a single TCP connection – is being standardized by IETF. This paper provides a comprehensive evaluation of the use of MPTCP to reduce latency and thus improve the quality of experience or QoE for cloud-based applications. In particular, the paper considers the possible reductions in latency that could be obtained by using MPTCP and multiple network paths between a cloud service and a mobile end user. To obtain an appreciation of the expected latency performance for different types of cloud traffic, three applications are studied, Netflix, Google Maps, and Google Docs, representing typical applications generating high-, mid-, and low-intensity traffic. The results suggest that MPTCP could provide significant latency reductions for cloud applications, especially for applications such as Netflix and Google Maps. Moreover, the results suggest that MPTCP offers a reduced latency despite a few percent packet loss, and in spite of limited differences in the round-trip times of the network paths in an MPTCP connection. Still, larger differences in the round-trip times seem to significantly increase the application latency, especially for Netflix, Google Maps, and similar applications. Thus, to become an even better alternative to these applications, this paper suggests that the MPTCP packet scheduling policy should be changed: Apart from the round-trip times of the network paths in a connection, it should also consider the difference in round-trip time between the network paths.

  • 50.
    Grinnemo, Karl-Johan
    et al.
    Karlstad University, Division for Information Technology.
    Brunström, Anna
    Karlstad University, Faculty of Economic Sciences, Communication and IT, Department of Computer Science.
    A Simulation Based Performance Evaluation of PRTP2002Report (Other academic)
    Abstract [en]

    PRTP is proposed to address the need of a transport service that is more suitable forapplications with soft real-time requirements, e.g., video broadcasting. It is an extensionfor partial reliability to TCP. The main idea behind PRTP is to exploit the fact thatmany soft real-time applications tolerate a limited amount of packet loss. In particular,PRTP enables an application to trade some of the reliability offered by TCP for improvedthroughput and interarrival jitter. This paper describes the design of PRTP andgives a detailed description of a simulation based performance evaluation. The performanceevaluation involved the performance of PRTP compared to TCP in long- as wellas in short-lived connections and showed that PRTP probably would give significantimprovements in performance, both in terms of throughput and interarrival jitter, for awide range of applications. The performance evaluation also suggested that PRTP is notTCP-friendly and altogether fair against competing flows and thus is not suitable for usein an environment where the main reason for packet loss is congestion, e.g., in a fixedInternet environment. We believe, however, that PRTP could be a viable alternative forwireless applications in error prone environments.

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