Network Traffic Control Design and Evaluation

Recently, the term bufferbloat has been coined to indicate the uncontrolled growth of the network queueing time. A number of network traffic control strategies have been proposed to control network queueing delay. Active Queue Management (AQM) algorithms such as RED, CoDel and PIE have been proposed to drop packets before the network queues become full and to notify upper layers, e.g., transport protocols, about possible congestion status. Innovative packet schedulers such as FQ-CoDel, have been introduced to prioritize flows which do not build queues. Strategies to reduce device buffering, e.g., BQL, have been proposed to increase the effectiveness of packet schedulers. Network experimentation through simulators such as ns-3, one of the most used network simulators, allows the study of bufferbloat and to evaluate solutions in a controlled environment. In this work, we aligned the ns-3 queueing system to the Linux one, one of the most used networking stacks. We introduced in ns-3 a traffic control module modelled after the Linux one. Our design allowed the introduction in ns-3 of schedulers such as FQ-CoDel and of algorithms to dynamically size the buffers such as BQL. Also, we devised a new emulation methodology to overcome some limitations and increase the emulation fidelity. Then, by using the new emulation methodology, we validated the traffic control module with its AQM algorithms (RED, CoDel, FQ-CoDel and PIE). Our experiments prove the high fidelity of network emulation and the high accuracy of the traffic control module and AQM algorithms. Then, we show two proposals of design and evaluation of traffic control strategies by using ns-3. Firstly, we designed and evaluated a traffic control layer for the backlog management in 3GPP stacks. The approach improves significantly the flows performance in LTE networks. Secondly, we highlighted possible design flaws in rate based AQM algorithms and proposed an alternative flow control approach. The approach allows the improvement of the effectiveness of AQM algorithms. Our work will allow researchers to design and evaluate in a more accurate manner traffic control strategies through ns-3 based simulation and emulation and to evaluate the accuracy of other modules implemented in ns-3

Buffer De-bloating in Wireless Access Networks

PhDExcessive buffering brings a new challenge into the networks which is known as Bufferbloat, which is harmful to delay sensitive applications. Wireless access networks consist of Wi-Fi and cellular networks. In the thesis, the performance of CoDel and RED are investigated in Wi-Fi networks with different types of traffic. Results show that CoDel and RED work well in Wi-Fi networks, due to the similarity of protocol structures of Wi-Fi and wired networks. It is difficult for RED to tune parameters in cellular networks because of the time-varying channel. CoDel needs modifications as it drops the first packet of queue and the head packet in cellular networks will be segmented. The major contribution of this thesis is that three new AQM algorithms tailored to cellular networks are proposed to alleviate large queuing delays. A channel quality aware AQM is proposed using the CQI. The proposed algorithm is tested with a single cell topology and simulation results show that the proposed algorithm reduces the average queuing delay for each user by 40% on average with TCP traffic compared to CoDel. A QoE aware AQM is proposed for VoIP traffic. Drops and delay are monitored and turned into QoE by mathematical models. The proposed algorithm is tested in NS3 and compared with CoDel, and it enhances the QoE of VoIP traffic and the average endto- end delay is reduced by more than 200 ms when multiple users with different CQI compete for the wireless channel. A random back-off AQM is proposed to alleviate the queuing delay created by video in cellular networks. The proposed algorithm monitors the play-out buffer and postpones the request of the next packet. The proposed algorithm is tested in various scenarios and it outperforms CoDel by 18% in controlling the average end-to-end delay when users have different channel conditions

Ruuhkan- ja jononhallinta tiedonsiirron alkuvaiheessa

Transmission Control Protocol (TCP) has served as the workhorse to transmit Internet traffic for several decades already. Its built-in congestion control mechanism has proved reliable to ensure the stability of the Internet, and congestion control algorithms borrowed from TCP are being applied largely also by other transport protocols. TCP congestion control has two main phases for increasing sending rate. Slow Start is responsible for starting up a flow by seeking the sending rate the flow should use. Congestion Avoidance then takes over to manage the sending rate for flows that last long enough. In addition, the flow is booted up by sending the Initial Window of packets prior to Slow Start. There is a large difference in the magnitude of sending rate increase during Slow Start and Congestion Avoidance. Slow Start increases the sending rate exponentially, whereas with Congestion Avoidance the increase is linear. If congestion is detected, a standard TCP sender reduces the sending rate heavily. It is well known that most of the Internet flows are short. It implies that flow startup is a rather frequent phenomenon.  Also, many traffic types exhibit an ON-OFF pattern with senders remaining idle for varying periods of time. As the flow startup under Slow Start causes exponential sending rate increase, the link load is often subject to exponential load transients that escalate in a few round trips into overload, if not controlled properly. It is true especially near the network edge where traffic aggregation is limited to a few users. Traditionally much of the congestion control research has focused on behavior during Congestion Avoidance and uses large aggregates during testing. To control router load, Active Queue Management (AQM) is recommended. The state-of-the-art AQM algorithms, however, are designed with little attention to Slow Start. This thesis focuses on congestion control and AQM during the flow startup. We explore what effect the Initial Window has to competing latency-sensitive traffic during a flow startup consisting of multiple parallel flows typical to Web traffic and investigate the impact of increasing Initial Window from three to ten TCP segments. We also highlight what the shortcomings are in the state-of-the-art AQM algorithms and formulate the challenges AQM algorithms must address to properly handle flow startup and exponential load transients. These challenges include the horizon problem, RTT (round-trip time) uncertainty and rapidly changing load. None of the existing AQM algorithms are prepared to handle these challenges. Therefore we explore whether an existing AQM algorithm called Random Early Detection (RED) can be altered to control exponential load transients effectively and propose necessary changes to RED. We also propose an entirely new AQM algorithm called Predict. It is the first AQM algorithm designed primarily for handling exponential load transients. Our evaluation shows that because of shortcomings in handling exponential load transients, the state-of-the-art AQM algorithms often respond too slowly or too fast depending on the actual RTT of the traffic. In contrast, the Predict AQM algorithm performs timely congestion indication without compromising throughput or latency unnecessarily, yielding low latency over a large range of RTTs. In addition, the load estimation in Predict is designed to be fully compatible with pacing and the timely congestion indication allows relaxing the large sending rate reduction on congestion detection.Ruuhkanhallinta on yksi keskeisimpiä Internetin sujuvan dataliikenteen turvaavia toimintoja. Ilman toimivaa ruuhkanhallintaa Internet ylikuormittuisi ja tulisi käyttökelvottomaksi. TCP-protokollaa (Transmission Control Protocol) on käytetty siirtämään tietoa Internetissä jo vuosia. Sen sisäänrakennettu ruuhkanhallinta on osoittautunut tehokkaaksi estämään pitkäkestoista verkon ylikuormittumista. TCP:n keskeiset ruuhkanhallinta-algoritmit ovat hidas aloitus (Slow Start) ja ruuhkan välttely (Congestion Avoidance). Hidasta aloitusta käytetään vallitsevan verkkotilanteen salliman lähetysnopeuden selvittämiseen. Kun sopiva lähetysnopeus on saavutettu, TCP siirtyy käyttämään ruuhkan välttely -algoritmia. Hitaan aloituksen ja ruuhkan välttelyn käyttämät algoritmit eroavat merkittävästi toisistaan. Hidas aloitus kasvattaa lähetysnopeutta eksponentiaalisesti, kun taas ruuhkan välttelyn aikana lähetysnopeus kasvaa lineaarisesti. Koska Internetissä yhden tiedonsiirtoyhteyden yli siirrettävä tietomäärä on tyypillisesti vähäinen ja uusien yhteyksien tiheä avaaminen sekä lähetystauot ovat erittäin yleisiä, aiheutuu hitaan aloituksen käyttämisestä usein toistuvia kuormapiikkejä, joiden aikana kuorman kasvu verkossa on eksponentiaalista. Verkon reuna-alueella lähellä käyttäjän laitetta verkon kuorma koostuu tyypillisesti vain yhden tai muutaman käyttäjän liikenteestä, jolloin kuormatason vaihtelu on suurta ja siirtyminen matalasta kuormasta ylikuormaan tapahtuu hitaan aloituksen käyttämisen takia erittäin nopeasti. Standardinmukaisen TCP:n hidas aloitus -algoritmi edellyttää ruuhkasignaalia joltakin verkkopolun liikennettä välittävältä reitittimeltä ennen kuin hidas aloitus lopetetaan ja siirrytään ruuhkan välttely -algoritmin käyttöön. Valtaosa aiemmasta ruuhkanhallintaan kohdistuvasta tietoliikennetutkimuksesta on sivuuttanut hitaan aloituksen ja on keskittynyt huomioimaan pelkästään ruuhkan välttely -algoritmille ominaisen toiminnan. Tämä väitöstyö sen sijaan keskittyy ongelmiin, joita aiheutuu uusien yhteyksien avaamisesta ja hitaan aloituksen käyttämisestä. Tämä väitöstyö osoittaa etteivät reitittimissä toimivat jononhallinta-algoritmit (Active Queue Management), jotka säätelevät ruuhkasignaaleja, yleensä reagoi riittävän nopeasti hitaaseen aloitukseen. Siksi hidas aloitus pääsee kiihdyttämään lähetysnopeutta huomattavasti yli verkkotilanteen mukaisesti määrittyvän sopivan lähetysnopeuden. Tämä ylitys siirtyy yleensä suoraan siirtoviiveeseen kasvattaen sitä, mikä myös osoitetaan tapahtuvan hitaan aloituksen aikana nykyaikaisia jononhallinta-algoritmeja käytettäessä. Toisaalta samat algoritmit voivat reagoida myös liian nopeasti, jos kiertoviive on pitkä, koska monet algoritmit perustuvat oletettuun maksimikiertoviiveeseen (yleensä 100 millisekunttia). Tämä väitöstyö valottaa kuinka jononhallinta-algoritmin on haasteellista selvittää todellinen kuormataso tutkimalla reitittimellä olevaa jonoa. Kyse on eräänlaisesta "horisontista", joka estää näkemästä kuormaa aiheuttavia tietoliikennepaketteja muilta verkkopolun osilta. Myös pidempikestoinen kuormamittaus on haasteellista, koska reitittimellä ei yleensä ole tietoa kuinka pitkää ajanjaksoa tulisi käyttää mittauksessa. Lisäksi hitaan aloituksen aikana nopeasti muuttuva kuormataso johtaa kuormamittaustulosten vanhenemiseen ennenaikaisesti. Jotta ruuhkasignaali voitaisiin lähettää ajoissa, täytyy jononhallinta-algoritmin huomioida kaikki kolme yllämainittua haastetta. Tämän väitöstyön osana suunniteltiin Predict-niminen jononhallinta-algoritmi, joka osaa havaita hitaan aloituksen ja reagoi siihen oikeaan aikaan estäen tehokkaasti hitaasta aloituksesta johtuvan ylikuormituksen ja siitä johtuvat suuret viivepiikit. Näin ollen Predict välttää muille algoritmeille ongelmalliset reagointinopeuden sudenkuopat. Lisäksi työssä selvitettiin kuinka uusien yhteyksien avaamista tulisi hallita, jotta samanaikaisesti mahdollisesti käynnissä olevalle interaktiiviselle tiedonsiirrolle ei aiheuteta ongelmia

Throughput and Delay on the Packet Switched Internet

The Internet has become a vital and essential part of modern everyday life. Services delivered by the Internet are used by people across the planet every moment of every day of the year. The Internet has proven a positive force for good improving the lives of billions of people worldwide. The power of the Internet to deliver this positive good to humanity relies on its ability to deliver life improving services. In my doctorate work culminating in this dissertation I have striven to sustain and increase the Internet's ability to deliver these services and to have a positive good effect upon humanity.The overarching purpose of this dissertation is to improve the Internet's ability to deliver life improving services. I have further divided this purpose into two goals. To improve the ability of applications operating in challenging network conditions to gain their fair share of the bandwidth resources and to reduce the delay with which these services are delivered. Every service delivered by the Internet consists of Internet objects that are delivered through communication paths across the Internet. The delivery of these objects is defined by the two characteristics; Throughput and delay. Throughput determines how much of an object can be delivered over a period of time and delay determines how long it takes to deliver an object.These two characteristics determine the Internet's ability to deliver objects across communication paths. Improving these two characteristics (bandwidth and delay) increase the ability of the Internet to deliver objects and thus improve the Internet's capability to deliver life improving services. To accomplish this goal I present projects along three areas of effort. These three areas of effort are: (1) Increase the ability of applications operating in challenging conditions to achieve their fair share of bandwidth. (2) Synthesize knowledge required to address the effort to reduce delay. (3) Develop protocols that reduce delay encountered in the communications paths of the Internet.In this dissertation I present projects along these three areas of effort that accomplish the two goals (increase bandwidth and reduce delay) to achieve the purpose of improving the Internet's ability to deliver essential and life improving services. These projects and their organization into areas of effort, goals and purpose are my contributions to the networking sciences

Congestion mitigation in LTE base stations using radio resource allocation techniques with TCP end to end transport

As of 2019, Long Term Evolution (LTE) is the chosen standard for most mobile and fixed wireless data communication. The next generation of standards known as 5G will encompass the Internet of Things (IoT) which will add more wireless devices to the network. Due to an exponential increase in the number of wireless subscriptions, in the next few years there is also an expected exponential increase in data traffic. Most of these devices will use Transmission Control Protocol (TCP) which is a type of network protocol for delivering internet data to users. Due to its reliability in delivering data payload to users and congestion management, TCP is the most common type of network protocol used. However, the ability for TCP to combat network congestion has certain limitations especially in a wireless network. This is due to wireless networks not being as reliable as fixed line networks for data delivery because of the use of last mile radio interface. LTE uses various error correction techniques for reliable data delivery over the air-interface. These cause other issues such as excessive latency and queuing in the base station leading to degradation in throughput for users and congestion in the network. Traditional methods of dealing with congestion such as tail-drop can be inefficient and cumbersome. Therefore, adequate congestion mitigation mechanisms are required. The LTE standard uses a technique to pre-empt network congestion by a mechanism known as Discard Timer. Additionally, there are other algorithms such as Random Early Detection (RED) that also are used for network congestion mitigation. However, these mechanisms rely on configured parameters and only work well within certain regions of operation. If the parameters are not set correctly then the TCP links can experience congestion collapse. In this thesis, the limitations of using existing LTE congestion mitigation mechanisms such as Discard Timer and RED have been explored. A different mechanism to analyse the effects of using control theory for congestion mitigation has been developed. Finally, congestion mitigation in LTE networks has been addresses using radio resource allocation techniques with non-cooperative game theory being an underlying mathematical framework. In doing so, two key end-to-end performance measurements considered for measuring congestion for the game theoretic models were identified which were the total end-to-end delay and the overall throughput of each individual TCP link. An end to end wireless simulator model with the radio access network using LTE and a TCP based backbone to the end server was developed using MATLAB. This simulator was used as a baseline for testing each of the congestion mitigation mechanisms. This thesis also provides a comparison and performance evaluation between the congestion mitigation models developed using existing techniques (such as Discard Timer and RED), control theory and game theory. As of 2019, Long Term Evolution (LTE) is the chosen standard for most mobile and fixed wireless data communication. The next generation of standards known as 5G will encompass the Internet of Things (IoT) which will add more wireless devices to the network. Due to an exponential increase in the number of wireless subscriptions, in the next few years there is also an expected exponential increase in data traffic. Most of these devices will use Transmission Control Protocol (TCP) which is a type of network protocol for delivering internet data to users. Due to its reliability in delivering data payload to users and congestion management, TCP is the most common type of network protocol used. However, the ability for TCP to combat network congestion has certain limitations especially in a wireless network. This is due to wireless networks not being as reliable as fixed line networks for data delivery because of the use of last mile radio interface. LTE uses various error correction techniques for reliable data delivery over the air-interface. These cause other issues such as excessive latency and queuing in the base station leading to degradation in throughput for users and congestion in the network. Traditional methods of dealing with congestion such as tail-drop can be inefficient and cumbersome. Therefore, adequate congestion mitigation mechanisms are required. The LTE standard uses a technique to pre-empt network congestion by a mechanism known as Discard Timer. Additionally, there are other algorithms such as Random Early Detection (RED) that also are used for network congestion mitigation. However, these mechanisms rely on configured parameters and only work well within certain regions of operation. If the parameters are not set correctly then the TCP links can experience congestion collapse. In this thesis, the limitations of using existing LTE congestion mitigation mechanisms such as Discard Timer and RED have been explored. A different mechanism to analyse the effects of using control theory for congestion mitigation has been developed. Finally, congestion mitigation in LTE networks has been addresses using radio resource allocation techniques with non-cooperative game theory being an underlying mathematical framework. In doing so, two key end-to-end performance measurements considered for measuring congestion for the game theoretic models were identified which were the total end-to-end delay and the overall throughput of each individual TCP link. An end to end wireless simulator model with the radio access network using LTE and a TCP based backbone to the end server was developed using MATLAB. This simulator was used as a baseline for testing each of the congestion mitigation mechanisms. This thesis also provides a comparison and performance evaluation between the congestion mitigation models developed using existing techniques (such as Discard Timer and RED), control theory and game theory

Accelerating Network Functions using Reconfigurable Hardware. Design and Validation of High Throughput and Low Latency Network Functions at the Access Edge

Providing Internet access to billions of people worldwide is one of the main technical challenges in the current decade. The Internet access edge connects each residential and mobile subscriber to this network and ensures a certain Quality of Service (QoS). However, the implementation of access edge functionality challenges Internet service providers: First, a good QoS must be provided to the subscribers, for example, high throughput and low latency. Second, the quick rollout of new technologies and functionality demands flexible configuration and programming possibilities of the network components; for example, the support of novel, use-case-specific network protocols. The functionality scope of an Internet access edge requires the use of programming concepts, such as Network Functions Virtualization (NFV). The drawback of NFV-based network functions is a significantly lowered resource efficiency due to the execution as software, commonly resulting in a lowered QoS compared to rigid hardware solutions. The usage of programmable hardware accelerators, named NFV offloading, helps to improve the QoS and flexibility of network function implementations. In this thesis, we design network functions on programmable hardware to improve the QoS and flexibility. First, we introduce the host bypassing concept for improved integration of hardware accelerators in computer systems, for example, in 5G radio access networks. This novel concept bypasses the system’s main memory and enables direct connectivity between the accelerator and network interface card. Our evaluations show an improved throughput and significantly lowered latency jitter for the presented approach. Second, we analyze different programmable hardware technologies for hardware-accelerated Internet subscriber handling, including three P4-programmable platforms and FPGAs. Our results demonstrate that all approaches have excellent performance and are suitable for Internet access creation. We present a fully-fledged User Plane Function (UPF) designed upon these concepts and test it in an end-to-end 5G standalone network as part of this contribution. Third, we analyze and demonstrate the usability of Active Queue Management (AQM) algorithms on programmable hardware as an expansion to the access edge. We show the feasibility of the CoDel AQM algorithm and discuss the challenges and constraints to be considered when limited hardware is used. The results show significant improvements in the QoS when the AQM algorithm is deployed on hardware. Last, we focus on network function benchmarking, which is crucial for understanding the behavior of implementations and their optimization, e.g., Internet access creation. For this, we introduce the load generation and measurement framework P4STA, benefiting from flexible software-based load generation and hardware-assisted measuring. Utilizing programmable network switches, we achieve a nanosecond time accuracy while generating test loads up to the available Ethernet link speed

Resilient and Scalable Forwarding for Software-Defined Networks with P4-Programmable Switches

Traditional networking devices support only fixed features and limited configurability. Network softwarization leverages programmable software and hardware platforms to remove those limitations. In this context the concept of programmable data planes allows directly to program the packet processing pipeline of networking devices and create custom control plane algorithms. This flexibility enables the design of novel networking mechanisms where the status quo struggles to meet high demands of next-generation networks like 5G, Internet of Things, cloud computing, and industry 4.0. P4 is the most popular technology to implement programmable data planes. However, programmable data planes, and in particular, the P4 technology, emerged only recently. Thus, P4 support for some well-established networking concepts is still lacking and several issues remain unsolved due to the different characteristics of programmable data planes in comparison to traditional networking. The research of this thesis focuses on two open issues of programmable data planes. First, it develops resilient and efficient forwarding mechanisms for the P4 data plane as there are no satisfying state of the art best practices yet. Second, it enables BIER in high-performance P4 data planes. BIER is a novel, scalable, and efficient transport mechanism for IP multicast traffic which has only very limited support of high-performance forwarding platforms yet. The main results of this thesis are published as 8 peer-reviewed and one post-publication peer-reviewed publication. The results cover the development of suitable resilience mechanisms for P4 data planes, the development and implementation of resilient BIER forwarding in P4, and the extensive evaluations of all developed and implemented mechanisms. Furthermore, the results contain a comprehensive P4 literature study. Two more peer-reviewed papers contain additional content that is not directly related to the main results. They implement congestion avoidance mechanisms in P4 and develop a scheduling concept to find cost-optimized load schedules based on day-ahead forecasts