Unicast UDP Usage Guidelines for Application Designers

Publisher PD

Estimating packet loss rate in the access through application-level measurements

End user monitoring of quality of experience is one of the necessary steps to achieve an effective and winning control over network neutrality. The involvement of the end user, however, requires the development of light and user-friendly tools that can be easily run at the application level with limited effort and network resources usage. In this paper, we propose a simple model to estimate packet loss rate perceived by a connection, by round trip time and TCP goodput samples collected at the application level. The model is derived from the well-known Mathis equation, which predicts the bandwidth of a steady-state TCP connection under random losses and delayed ACKs and it is evaluated in a testbed environment under a wide range of different conditions. Experiments are also run on real access networks. We plan to use the model to analyze the results collected by the "network neutrality bot" (Neubot), a research tool that performs application-level network-performance measurements. However, the methodology is easily portable and can be interesting for basically any user application that performs large downloads or uploads and requires to estimate access network quality and its variation

Transfer Control for Resilient End-to-End Transport

Residing between the network layer and the application layer, the transport layer exchanges application data using the services provided by the network. Given the unreliable nature of the underlying network, reliable data transfer has become one of the key requirements for those transport-layer protocols such as TCP. Studying the various mechanisms developed for TCP to increase the correctness of data transmission while fully utilizing the network's bandwidth provides us a strong background for our study and development of our own resilient end-to-end transport protocol. Given this motivation, in this thesis, we study the different TCP's error control and congestion control techniques by simulating them under different network scenarios using ns-3. For error control, we narrow our research to acknowledgement methods such as cumulative ACK - the traditional TCP's way of ACKing, SACK, NAK, and SNACK. The congestion control analysis covers some TCP variants including Tahoe, Reno, NewReno, Vegas, Westwood, Westwood+, and TCP SACK

Enabling Multipath and Multicast Data Transmission in Legacy and Future Internet

The quickly growing community of Internet users is requesting multiple applications and services. At the same time the structure of the network is changing. From the performance point of view, there is a tight interplay between the application and the network design. The network must be constructed to provide an adequate performance of the target application. In this thesis we consider how to improve the quality of users' experience concentrating on two popular and resource-consuming applications: bulk data transfer and real-time video streaming. We share our view on the techniques which enable feasibility and deployability of the network functionality leading to unquestionable performance improvement for the corresponding applications. Modern mobile devices, equipped with several network interfaces, as well as multihomed residential Internet hosts are capable of maintaining multiple simultaneous attachments to the network. We propose to enable simultaneous multipath data transmission in order to increase throughput and speed up such bandwidth-demanding applications as, for example, file download. We design an extension for Host Identity Protocol (mHIP), and propose a multipath data scheduling solution on a wedge layer between IP and transport, which effectively distributes packets from a TCP connection over available paths. We support our protocol with a congestion control scheme and prove its ability to compete in a friendly manner against the legacy network protocols. Moreover, applying game-theoretic analytical modelling we investigate how the multihomed HIP multipath-enabled hosts coexist in the shared network. The number of real-time applications grows quickly. Efficient and reliable transport of multimedia content is a critical issue of today's IP network design. In this thesis we solve scalability issues of the multicast dissemination trees controlled by the hybrid error correction. We propose a scalable multicast architecture for potentially large overlay networks. Our techniques address suboptimality of the adaptive hybrid error correction (AHEC) scheme in the multicast scenarios. A hierarchical multi-stage multicast tree topology is constructed in order to improve the performance of AHEC and guarantee QoS for the multicast clients. We choose an evolutionary networking approach that has the potential to lower the required resources for multimedia applications by utilizing the error-correction domain separation paradigm in combination with selective insertion of the supplementary data from parallel networks, when the corresponding content is available. Clearly both multipath data transmission and multicast content dissemination are the future Internet trends. We study multiple problems related to the deployment of these methods.Internetin nopeasti kasvava käyttäjäkunta vaatii verkolta yhä enemmän sovelluksia ja palveluita. Samaan aikaan verkon rakenne muuttuu. Suorituskyvyn näkökulmasta on olemassa selvä vuorovaikutussovellusten ja verkon suunnittelun välillä. Verkko on rakennettava siten, että se pystyy takaamaan riittävän suorituskyvyn halutuille palveluille. Tässä väitöskirjassa pohditaan, miten verkon käyttökokemusta voidaan parantaa keskittyen kahteen suosittuun ja resursseja vaativaan sovellukseen: tiedonsiirtoon ja reaaliaikaiseen videon suoratoistoon. Esitämme näkemyksemme tekniikoista, jotka mahdollistavat tarvittavien verkkotoiminnallisuuksien helpon toteuttavuuden sekä kiistatta parantavat sovelluksien suorityskykyä. Nykyaikaiset mobiililaitteet monine verkkoyhteyksineen, kuten myös kotitietokoneet, pystyvät ylläpitämään monta internet-yhteyttä samanaikaisesti. Siksi ehdotamme monikanavaisen tiedonsiirron käyttöä suorituskyvyn parantamiseksi ja etenkin vaativien verkkosovelluksien, kuten tiedostonsiirron, nopeuttamiseksi. Tässä väitöskirjassa suunnitellaan Host Identity Protocol (mHIP) -laajennus, sekä esitetään tiedonsiirron vuorotteluratkaisu, joka hajauttaa TCP-yhteyden tiedonsiirtopaketit käytettävissä oleville kanaville. Protokollamme tueksi luomme myös ruuhkautumishallinta-algoritmin ja näytämme sen pystyvän toimimaan yhteen nykyisien verkkoprotokollien kanssa. Tämän lisäksi tutkimme peliteoreettista mallinnusta käyttäen, miten monikanavaiset HIP-verkkopäätteet toimivat muiden kanssa jaetuissa verkoissa. Reaaliaikaisten sovellusten määrä kasvaa nopeasti. Tehokas ja luotettava multimediasisällön siirto on olennainen vaatimus nykypäivän IP-verkoissa. Tässä työssä ratkaistaan monilähetyksen (multicast) jakelustruktuurin skaalautuvuuteen liittyviä ongelmia. Ehdotamme skaalautuvaa monilähetysarkkitehtuuria suurille peiteverkoille. Ratkaisumme puuttuu adaptiivisen virhekorjauksen (Adaptive Hybrid Error Correction, AHEC) alioptimaalisuuteen monilähetystilanteissa. Luomme hierarkisen monivaiheisen monilähetyspuutopologian parantaaksemme AHECin suorituskykyä, sekä taataksemme monilähetysasiakkaiden palvelun laadun. Valitsimme evoluutiomaisen lähestymistavan, jolla on potentiaalia keventää multimediasovelluksien verkkoresurssivaatimuksia erottamalla virhekorjauksen omaksi verkkotunnuksekseen, sekä käyttämällä valikoivaa täydentävää tiedonlisäystä rinnakkaisverkoista vastaavan sisällön ollessa saatavilla. Sekä monikanava- että monilähetystiedonsiirto ovat selvästi osa internetin kehityssuuntaa. Tässä väitöskirjassa tutkimme monia ongelmia näiden tekniikoiden käyttöönottoon liittyen

End-to-End Resilience Mechanisms for Network Transport Protocols

The universal reliance on and hence the need for resilience in network communications has been well established. Current transport protocols are designed to provide fixed mechanisms for error remediation (if any), using techniques such as ARQ, and offer little or no adaptability to underlying network conditions, or to different sets of application requirements. The ubiquitous TCP transport protocol makes too many assumptions about underlying layers to provide resilient end-to-end service in all network scenarios, especially those which include significant heterogeneity. Additionally the properties of reliability, performability, availability, dependability, and survivability are not explicitly addressed in the design, so there is no support for resilience. This dissertation presents considerations which must be taken in designing new resilience mechanisms for future transport protocols to meet service requirements in the face of various attacks and challenges. The primary mechanisms addressed include diverse end-to-end paths, and multi-mode operation for changing network conditions

Evaluating and improving the performance of video content distribution in lossy networks

The contributions in this research are split in to three distinct, but related, areas. The focus of the work is based on improving the efficiency of video content distribution in the networks that are liable to packet loss, such as the Internet. Initially, the benefits and limitations of content distribution using Forward Error Correction (FEC) in conjunction with the Transmission Control Protocol (TCP) is presented. Since added FEC can be used to reduce the number of retransmissions, the requirement for TCP to deal with any losses is greatly reduced. When real-time applications are needed, delay must be kept to a minimum, and retransmissions not desirable. A balance, therefore, between additional bandwidth and delays due to retransmissions must be struck. This is followed by the proposal of a hybrid transport, specifically for H.264 encoded video, as a compromise between the delay-prone TCP and the loss-prone UDP. It is argued that the playback quality at the receiver often need not be 100% perfect, providing a certain level is assured. Reliable TCP is used to transmit and guarantee delivery of the most important packets. The delay associated with the proposal is measured, and the potential for use as an alternative to the conventional methods of transporting video by either TCP or UDP alone is demonstrated. Finally, a new objective measurement is investigated for assessing the playback quality of video transported using TCP. A new metric is defined to characterise the quality of playback in terms of its continuity. Using packet traces generated from real TCP connections in a lossy environment, simulating the playback of a video is possible, whilst monitoring buffer behaviour to calculate pause intensity values. Subjective tests are conducted to verify the effectiveness of the metric introduced and show that the results of objective and subjective scores made are closely correlated