L'évolution des réseaux actuels vise principalement l'amélioration des performances à travers une augmentation importante des débits disponibles. Cependant, les besoins des applications multimédia se situent également dans l'accès à de nouvelles fonctionnalités telles que le multipoint et la garantie de qualité de service (QoS). Pour satisfaire ces besoins, nous proposons d'associer une architecture pour le multipoint avec QoS aux technologies de commutation. Une première instanciation de cette approche nous a amenés à étudier l'intégration de RSVP sur les réseaux ATM, et à proposer une solution palliant l'absence de service multipoint avec QoS capable de soutenir efficacement RSVP. Nous nous intéressons ainsi à CLIP, l'une des techniques d'intégration d'IP sur ATM, pour réaliser l'intégration du modèle RSVP. La contribution repose sur l'utilisation de plusieurs arbres multipoints pour éviter les limitations intrinsèques aux réseaux ATM. Cette approche a été expérimentée sur la plate-forme MIRIHADE à l'aide d'une application vidéo à codage hiérarchique. La commutation associée à une signalisation orientée connexion se prêtant mal à l'intégration des nouvelles fonctionnalités, nous nous sommes orientés vers l'utilisation d'une signalisation en mode non connecté à travers une solution de type ``Commutation par label''. Appliquée à RSVP, nous avons intégré directement la signalisation de la commutation dans celle assurant les nouvelles fonctionnalités pour permettre la commutation directe du trafic de données tout en conservant les fonctionnalités de la couche réseau. Notre solution, appelée ``RSVP Switching'', est en cours d'expérimentation sur la plate-forme SAFIR, pour un projet de simulation interactive distribuée ayant des besoins importants en terme de multipoint avec QoS.

Enda sedan den först introducerades på 1870-talet har telefonen spelat en viktigt roll i kommunikationen människor mellan. Utvecklingen inom telekombranschen har skett i ett rasande snabbt tempo under de senaste 30-40 år, och idag är det vanligare med en mobiltelefon än en fast telefon. En populär variant av mobiltelefonen är den så kallade smarta mobilen som bland annat gör det möjligt att nyttja mer avancerade program och tjänster än i en traditionell mobiltelefon. Sedan början av 2000-talet har röstsamtal över internet, så kallad VoIP, ökat allt mer. Fördelen med VoIP är främst att samtal mellan datorer i regel är helt kostnadsfria vilket gör dessa tjänster utmärkta för långdistanssamtal.   I och med att antalet smarta mobiler ökar samtidigt som marknaden för VoIP växer har mobiloperatörerna börjat se detta som ett större hot mot sina intäkter. Idag har i princip alla mobiloperatörer i Sverige ett förbud mot VoIP i sina så kallade surfabonnemang. Detta innebär att slutanvändarna hindras att nyttja valfria tjänster samtidigt som tjänsteutvecklare får det svårt att ta sig in på marknaden.   I slutet av 2009 antog Europaparlamentet två så kallade ändringsdirektiv som innebär att regelverket som berör elektroniska kommunikationer inom EU moderniseras. Samtliga medlemsstater hade därefter 18 månader på sig att implementera de nya bestämmelserna i sina respektive lagstiftningar. I Sverige trädde ett flertal bestämmelser i kraft den 1 juli 2011, bland dessa fanns 5 kap 6 d § LEK som introducerar det nya begreppet lägsta tjänstekvalitet.   Syftet med denna uppsats är att utifrån nätneutralitetsbegreppet utreda huruvida den nya bestämmelsen, 5 kap 6 d § LEK, kommer att leda till att förbudet mot VoIP i mobiloperatörernas nät hävs.   Utifrån det material som funnits tillgängligt har följande slutsats kunnat göras. Det går idag inte till fullo att veta huruvida bestämmelsen kommer att häva VoIP-förbudet, detta då den varken använts av PTS eller prövats i ett överklagande av berörd part.

Technological advancements have led to an increase in the number of wireless and mobile devices such as PDAs, laptops and smart phones. This has resulted in an ever- increasing demand for wireless access to the Internet. Hence, wireless mobile traffic is expected to form a significant fraction of Internet traffic in the near future, over the so-called Mobile Internet Protocol (MIP) networks. For real-time applications, such as voice, video and process monitoring and control, deployed over standard IP networks, network resources must be properly allocated so that the mobile end-user is guaranteed a certain Quality of Service (QoS). As with the wired and fixed IP networks, MIP networks do not offer any QoS guarantees. Such networks have been designed for non-real-time applications. In attempts to deploy real-time applications in such networks without requiring major network infrastructure modifications, the end-points must provide some level of QoS guarantees. Such QoS guarantees or QoS control, requires ability of predictive capabilities of the end-to-end flow characteristics. In this research network flow accumulation is used as a measure of end-to-end network congestion. Careful analysis and study of the flow accumulation signal shows that it has long-term dependencies and it is very noisy, thus making it very difficult to predict. Hence, this work predicts the moving average of the flow accumulation signal. Both single-step and multi-step predictors are developed using linear system identification techniques. A multi-step prediction error of up to 17% is achieved for prediction horizon of up to 0.5sec. The main thrust of this research is on the impact of wireless losses on the ability to predict end-to-end flow accumulation. As opposed to wired, congestion related packet losses, the losses occurring in a wireless channel are to a large extent random, making the prediction of flow accumulation more challenging. Flow accumulation prediction studies in this research demonstrate that, if an accurate predictor is employed, the increase in prediction error is up to 170% when wireless loss reaches as high as 15% , as compared to the case of no wireless loss. As the predictor accuracy in the case of no wireless loss deteriorates, the impact of wireless losses on the flow accumulation prediction error decreases.

<p>På några år har användningen av IP-telefoni inom svenska företag tredubblats och vi börjar nu se ett attitydskifte gentemot en teknik som länge dragits med dåligt rykte på grund av kvalitetsbrister.</p><p>IP-telefoni är ett resultat av att data- och telefonikommunikation vävs samman och att kapaciteten i nätverken ökar. Detta medför att man genom att införa IP-telefoni kan genomföra kostnadsbesparingar och effektivisering. Men tekniken erbjuder även ökad funktionalitet och därmed även mervärde för organisationer som använder IP-telefoni. Inser organisationer detta och utnyttjar man fördelarna med tekniken för att skapa acceptans inom organisationen?</p><p>Syftet är att undersöka vilka faktorer som motiverar en övergång från kretskopplad telefoni till IP-telefoni, och hur man inom en organisation skapar acceptans för den nya tekniken.</p><p>Syftet har uppfyllts genom en kvalitativ studie som byggt på semi-standardiserade intervjuer. Studien genomfördes hos fyra svenska organisationer som implementerat en ”ren” IP-telefonilösning med IP-växel och hårdvarutelefoner.</p><p>Resultatet av studien visar att det är vanligt att utnyttja en större händelse som startskott för implementationen. Detta kan röra sig om en flytt eller en renovering av organisationens lokaler. De främsta orsakerna till att man väljer IP-telefoni har visat sig vara kostnadsbesparingar, effektivisering av nätverksinfrastrukturen samt att man ser tekniken som en investering för framtiden. Det har visat sig att de företag som genomfört en grundlig studie av IP-telefoni som teknik, inledningsvis har identifierat fler fördelar med tekniken.</p><p>Vidare har studien även visat att organisationer upplever ett initialt motstånd till förändringen från anställda och att detta motstånd ofta uppkommer i anknytning till tekniska problem som uppstår i samband med implementationen.</p><p>Det är mycket ovanligt att organisationer har en speciell fas för att skapa acceptans för förändringen, vilket troligtvis beror på att man ser användandet som liknande med traditionell PSTN-telefoni och att resurser istället läggs på att lösa tekniska problem.</p> / <p>Over the last few years there has been a threefold increase in the use of IP telephony within Swedish companies. We are now beginning to see a change in attitude towards a technology that thanks to poor Quality of Service has been struggling with a bad reputation. IP telephony is a result of computer and telephone networks being con-verged, and by doing this, costs can be saved and efficiency boosted. But the technology does also offer increased functionality and thereby added value to the organisations using it. But do organisations realize this, and do they exploit the benefits of the technology in order to create acceptance for it within the organisation?</p><p>The purpose is to examine what the motivating factors are for a transition from PSTN telephony to IP telephony, and also to study how acceptance for the new technology is created within the organisation.</p><p>The purpose has been fulfilled by conducting a qualitative study based on semi-standardized interviews. The study has been conducted at four Swedish organisations that have implemented a "pure" IP telephony solution including IP gateways and IP telephones.</p><p>The result of the study shows that it is common to use a "big event" as a starting block for the implementation. This could be a relocation or the restoration of an office building. The main reason for choosing IP telephony is the fact that convergence cuts costs, boosts efficiency, and that organisations see IP telephony as an investment for the future. We have noticed that organisations that conduct a thorough study of IP telephony have identified more functional advantages with the technology.</p><p>Furthermore, the study has shown that organisations experience initial resistance from users, and that this resistance often arises in connection to technical problems that transpire during the implementation.</p><p>It is rare for organisations to have special phase for creation of acceptance, which most likely is due to the fact that organisations see the use of IP telephony as similar to the use of traditional PSTN telephony, and that resources instead are being placed on solving technical problems</p>

Αντικείμενο της παρούσας Διατριβής είναι η μελέτη της απόδοσης και η εισαγωγή νέων χαρακτηριστικών σε μοντέλα για την παροχή υπηρεσιών με εγγυήσεις ποιότητας στα σύγχρονα IP δίκτυα καθώς και η εισαγωγή των απαραίτητων επιχειρησιακών λειτουργιών για την εφαρμογή των μοντέλων αυτών, με στόχο τη βελτίωση της απόδοσης. Ακολουθώντας μια καταγραφή και αξιολόγηση των μηχανισμών και αρχιτεκτονικών που εισάγουν τη διαφοροποίηση εξυπηρέτησης στα IP δίκτυα, η μελέτη που παρουσιάζεται εδώ ακολουθεί σε όλα της τα στάδια τις αρχές της αρχιτεκτονικής DiffServ, η οποία επιτρέπει την παροχή ενός συγκεκριμένου εύρους υπηρεσιών με εγγυήσεις ποιότητας σε συναθροίσεις ροών και περιορίζει την πολυπλοκότητα στα όρια του δικτύου. Η απόδοση και η αποτελεσματικότητα των μηχανισμών και λειτουργιών διαφοροποίησης εξυπηρέτησης με βάση την αρχιτεκτονική DiffServ αξιολογούνται με βάση την εξασφάλιση εγγυημένης χωρητικότητας, φραγμένης από-άκρο-σε-άκρο καθυστέρησης, ελαχιστοποιημένων απωλειών πακέτων και φραγμένου jitter. Ωστόσο, σε κάθε μοντέλο υπηρεσίας για την παροχή QoS, τονίζεται η σημασία της εξασφάλισης των καλύτερων δυνατών εγγυήσεων ποιότητας με την ελάχιστη δυνατή πολυπλοκότητα. Τα διαφορετικά μοντέλα υπηρεσιών θέτουν συγκεκριμένους στόχους λαμβάνοντας υπόψη τις ανάγκες των εφαρμογών στις οποίες απευθύνονται. Οι δύο επικρατέστερες κατηγορίες υπηρεσιών στα πλαίσια της αρχιτεκτονικής DiffServ είναι η κατηγορία των υπηρεσιών μέγιστης προτεραιότητας και η κατηγορία των υπηρεσιών εγγυημένης χωρητικότητας σε συνθήκες συμφόρησης. Στην πρώτη κατηγορία, προτείνεται το μοντέλο υπηρεσίας Gold, το οποίο ακολουθεί τις αρχές τις αρχιτεκτονικής DiffServ για να παρέχει βέλτιστη ποιότητα εξυπηρέτησης σε συναθροίσεις IP ροών, ενώ ταυτόχρονα μπορεί να εφαρμοστεί πρακτικά σε δίκτυα παραγωγής. Στη δεύτερη κατηγορία, προτείνεται το μοντέλο υπηρεσίας Relative για την παροχή υπηρεσιών εγγυημένης χωρητικότητας σε συνθήκες συμφόρησης, με βασικά χαρακτηριστικά την μείωση της υπολογιστικής πολυπλοκότητας και την βελτίωση της δικαιοσύνης μεταξύ των εξυπηρετούμενων TCP ροών. Η υπηρεσία Gold διατηρεί την αρχή της επεκτασιμότητας και παρέχει αυστηρές εγγυήσεις ποιότητας αλλά ταυτόχρονα επιτρέπει την εφαρμογή μηχανισμού ελέγχου αποδοχής νέων αιτημάτων χωρίς διακοπή της λειτουργίας του δικτύου. Eισάγει επίσης ένα νέο χαρακτηριστικό σε σχέση με τα υπάρχοντα σχήματα: την διαφοροποίηση ως προς την εγγυημένη μέγιστη καθυστέρηση που παρέχεται στις ροές. Υλοποιείται με τη χρονοδρομολόγηση LA-EDF, που εισάγει την έννοια της διαφοροποιημένης εξυπηρέτησης εντός της ίδιας κλάσης υπηρεσίας και λειτουργεί ως υποστηρικτικός μηχανισμός του ελέγχου αποδοχής κλήσεων, τον αλγόριθμο DBAC για την αποδοχή κλήσεων χωρίς επέμβαση στη λειτουργία του δικτύου και την δρομολόγηση εξισορρόπησης φόρτου για την καλύτερη αξιοποίηση των διαθέσιμων πόρων χωρίς να παραβιάζονται οι εγγυήσεις ποιότητας. Η υπηρεσία Relative επιτυγχάνει προσαρμοστικότητα σε συνθήκες μεταβαλλόμενου φόρτου, δίκαιη διαφοροποίηση, υψηλή απόδοση, αύξηση της χρησιμοποίησης των διαθέσιμων πόρων ενώ αντιμετωπίζει πολλές από τις αδυναμίες που παρουσιάζουν αντίστοιχα μοντέλα. Υλοποιείται με τον μηχανισμό μαρκαρίσματος TWAM ο οποίος εφαρμόζεται στο σημείο εισόδου των ροών στο δίκτυο και αντιμετωπίζει τα θέματα της μη δίκαιης μεταχείρισης TCP ροών με τη μικρότερη δυνατή υπολογιστική επιβάρυνση σε σχέση με υπάρχοντες μηχανισμούς μαρκαρίσματος και τον μηχανισμό DWRED για την ενεργητική διαχείριση του αποθηκευτικού χώρου των ουρών, ο οποίος λειτουργεί με βάση το μαρκάρισμα της κίνησης που επιφέρει ο TWAM και προσαρμόζεται στις μεταβαλλόμενες συνθήκες λειτουργίας. Μεταξύ των καθοριστικών παραγόντων για την ευρεία υιοθέτηση υπηρεσιών βασισμένων στην αρχιτεκτονική DiffServ στα σύγχρονα δίκτυα παραγωγής, αναδεικνύονται η εισαγωγή ευέλικτων επιχειρηματικών μοντέλων για την υλοποίηση των υπηρεσιών αυτών στο εσωτερικό ενός δικτύου καθώς και χρέωσης των παρεχόμενων υπηρεσιών. Ο ορισμός Συμβολαίων Εξασφάλισης Επιπέδου Υπηρεσιών (ΣΕΕΥ) για δίκτυα που υποστηρίζουν την παροχή QoS υπηρεσιών με βάση την αρχιτεκτονική DiffServ έχει ως στόχο την εξασφάλιση της συμβατότητας των παρεχόμενων από διαφορετικά διασυνδεδεμένα δίκτυα υπηρεσιών προκειμένου για την από-άκρο-σε-άκρο εξασφάλιση εγγυήσεων ποιότητας. Προτείνεται ένα πρότυπο για την υλοποίηση διμερών ΣΕΕΥ σε IP δίκτυα που παρέχουν υπηρεσίες μέγιστης προτεραιότητας καθώς και μια μεθοδολογία για την υλοποίηση από-άκρο-σε-άκρο Συμβολαίων πάνω από διασυνδεδεμένα δίκτυα. Σε ένα IP δίκτυο, η εισαγωγή ενός αριθμού κλάσεων υπηρεσιών που διαφοροποιούνται στις παρεχόμενες ποιοτικές εγγυήσεις απαιτεί την εισαγωγή διαφοροποιημένων μοντέλων χρέωσης που επιπρόσθετα οδηγούν τους χρήστες στην επιλογή της κατάλληλης κλάσης υπηρεσίας η οποία μεγιστοποιεί την αντιληπτή χρησιμότητα. Προκειμένου για τη χρέωση υπηρεσιών με βάση την αρχιτεκτονική DiffServ, τα προφίλ κίνησης των χρηστών και οι διαφορές στην αντιληπτή ποιότητα αντιπροσωπεύουν τη χρησιμότητα που αντιλαμβάνεται ο χρήστης. Προτείνεται ένα μοντέλο χρέωσης όπου το προφίλ της κίνησης αποτελεί το αντικείμενο διαπραγμάτευσης του χρήστη με τον πάροχο, αφού ο χρήστης συνυπολογίσει τις εγγυήσεις ποιότητας εξυπηρέτησης που ανακοινώνονται από τον τελευταίο προκαταβολικά. Η καινοτομία του προτεινόμενου μοντέλου συνίσταται στις εξωτερικές συνθήκες (externalities) που υπεισέρχονται στα υφιστάμενα κόστη και προκαλούνται από τη φύση των υπηρεσιών που υλοποιούνται με βάση το μοντέλο DiffServ, καθώς επίσης και στον καθορισμό των πραγματικών τιμών με βάση τις οποίες χρεώνονται οι χρήστες. / The goal of this Dissertation is to study the performance of existing tools and the introduction of new features to quality of service provisioning models in IP networks as well as the introduction of the business models required for applying these models in an operational environment in ways that the performance is improved. Following the evaluation of mechanisms and architectures for differentiation of service in IP networks, the principles of the DiffServ framework have been adopted. The DiffServ framework specifies the provision of a set of services with qualitative guarantees to traffic aggregates, while keeping complexity at the network edges. The performance and effectiveness of service differentiation mechanisms according to the principles of the DiffServ framework are evaluated according to the following metrics: guaranteed capacity, bounded end-to-end delay, minimization of packet losses and jitter. However, in any QoS model, it is important to ensure the best quality possible by keeping complexity low. Each QoS model is designed to meet the needs of a different traffic type. The two prevailing service models within the DiffServ framework are the maximum priority, maximum quality model and the guaranteed capacity under congestion model. The proposed Gold service falls within the first category above, offering advanced quality to IP traffic aggregates with a set of principles that can easily be applied to operational networks. The proposed Relative service model provides guaranteed capacity under congestion by reducing the complexity and improving fairness among TCP flows. The Gold service preserves scalability and provides strict quality guarantees, incorporating a call admission control mechanism that operates without interfering with the network operations. It introduces a novel feature: differentiation of the guarantees on end-to-end delay provided to traffic flow. It is implemented using LA-EDF scheduling that introduces service differentiation within the same class and supports the call admission control functions, the DBAC algorithm for admission control and flow routing with load balancing for optimizing the use of available resources without compromising in terms of the guaranteed quality. The Relative service achieves high adaptability in transient load conditions, fair differentiation, high quality, increase in the utilization of available resources without demonstrating the same weaknesses as equivalent service models. It is implemented using the TWAM marking mechanism, which is applied at the network ingress and ensures fairness with less overhead than similar mechanisms, and DWRED, the active queue management mechanism that depends upon the TWAM marking and adapts to the varying load levels. The introduction of effective business and pricing models is crucial for the adoption of qualitative service models based on the DiffServ framework in a production network. The definition of Service Level Agreements (SLAs) for networks that provide QoS according to the principles of the DiffServ framework aims at introducing compatibility among the services provided for the provisioning of end-to-end quality guarantees. A template for the implementation for bilateral SLAs between networks that support the maximum priority, maximum quality service model is proposed, together with a methodology for implementing, based on the bilateral SLA, an end-to-end SLA over multiple domains. In an IP network, the introduction of a set of services classes with differing quality guarantees necessitates the application of differentiated pricing models that lead the users to the selection of the appropriate service class in order to maximize their perceived utility. Based on the principles of the DiffServ framework, the utility for each user is determined by the profile of his traffic and the quality of service he perceives. The proposed pricing model appoints the traffic profile as the parameter for negotiation between the user and the provider, after the user assesses the quality guarantees announced by the provider prior to the service provisioning. The innovation here lies in the introduction of externalities to the costs induced as well as the announcement of the actual prices upon which the user will eventually be charged. The externalities are imposed by the nature of the service models implemented according to the DiffServ framework.

Κεντρικό αντικείμενο αυτής της Διδακτορικής Διατριβής αποτελεί η μελέτη του συνδυασμού δύο εκ των βασικότερων εξελίξεων που σχετίζονται με το επίπεδο του IP πρωτοκόλλου στο Internet: της δυνατότητας για την παροχή εγγυήσεων ποιότητας (Quality of Service) σε τμήμα της συνολικής κίνησης που διακινείται μέσα από τα IP δίκτυα, καθώς και της ανάγκης αναβάθμισης του IPv4 πρωτοκόλλου στο IPv6. Επίσης αντικείμενο της παρούσας εργασίας είναι η ανάπτυξη μηχανισμών και αλγορίθμων για την αποδοτική διαχείριση των πόρων, τον όσο το δυνατόν δίκαιο καταμερισμό της ποιότητας υπηρεσίας, καθώς και τη δυνατότητα συνεργασίας και διαλειτουργικότητας μεταξύ διαφορετικών αυτόνομων δικτυακών τμημάτων με αυτοματοποιημένο τρόπο (χωρίς δηλαδή να χρειάζεται η παρέμβαση ενός ανθρώπου διαχειριστή στις περισσότερες περιπτώσεις). Για το σκοπό αυτό έχουν προταθεί διάφορες προσεγγίσεις όσον αφορά μεσίτες εύρους ζώνης, οι οποίες μελετώνται στην εργασία αυτή, ενώ προτείνονται αλγόριθμοι και μηχανισμοί για τη βελτίωση της λειτουργίας και της απόδοσής τους. Το IPv4 είχε τη δυνατότητα υλοποίησης μηχανισμών QoS στο επίπεδο δικτύου με τη χρήση του πεδίου TOS (Type Of Service). Στην πράξη όμως το πεδίο αυτό έμεινε σε μεγάλο βαθμό ανεκμετάλλευτο. Το IPv6 επεκτείνει και βελτιώνει την ιδέα αυτή, παρέχοντας δύο νέα πεδία στην στάνταρ επικεφαλίδα, τα Traffic Class και Flow Label, τα οποία μπορούν να χρησιμοποιηθούν προς αυτήν την κατεύθυνση. Η χρήση των πεδίων αυτών, όπως και γενικότερα η χρήση του IPv6 βρίσκονται ακόμα σε πειραματικό επίπεδο. Καθώς όμως το IPv6 περνάει σιγά-σιγά στο προσκήνιο και ετοιμάζεται να υποκαταστήσει το κυρίαρχο έως τώρα IPv4, παρουσιάζει ιδιαίτερο ενδιαφέρον η διερεύνηση του τρόπου με τον οποίο θα αξιοποιηθούν πρακτικά οι QoS δυνατότητες που προσφέρει το IPv6. Μία σημαντική παράμετρος της υποστήριξης QoS Μηχανισμών από άκρο σε άκρο είναι η συνεργασία μεταξύ διαφορετικών αυτόνομων τμημάτων (domains) που απαιτείται προκειμένου η κίνηση να υφίσταται προνομιακή μεταχείριση καθ’ όλη τη διαδρομή της και να της παρέχονται οι αναγκαίες εγγυήσεις ποιότητας. Η διαπραγμάτευση της συνεργασίας αυτής είναι σαφές ότι πρέπει να είναι όσο το δυνατόν αυτοματοποιημένη για να μπορούν τέτοιου είδους υπηρεσίες να γνωρίσουν ευρύτερη διάδοση. Για το σκοπό αυτό έχει από το RFC 2638 της IETF οριστεί η μονάδα του Bandwidth Broker (μεσίτης εύρους ζώνης). Ελέγχει το δικτυακό φόρτο αποδεχόμενη ή απορρίπτοντας αιτήματα για συγκεκριμένο bandwidth με εγγυήσεις QoS. Οι Bandwidth Brokers χρειάζεται να εγκαθιδρύσουν σχέσεις περιορισμένης εμπιστοσύνης με τις αντίστοιχες μονάδες στα γειτονικά domains, αντίθετα με άλλες αρχιτεκτονικές που απαιτούν τον καθορισμό των χαρακτηριστικών μιας ροής στους δρομολογητές κατά μήκος του από άκρο σε άκρο μονοπατιού. Επομένως η αρχιτεκτονική του Bandwidth Broker δίνει τη δυνατότητα να κρατηθεί η πληροφορία στο επίπεδο του διαχειριστικού domain, αντί να πρέπει να κρατηθεί σε κάθε δρομολογητή, και η DiffServ αρχιτεκτονική δίνει τη δυνατότητα να περιοριστεί η πληροφορία αυτή μόνο για τους ακραίους δρομολογητές κάθε domain. Στα πλαίσια της εργασίας αυτής ασχοληθήκαμε επίσης με τη μονάδα ελέγχου αποδοχής ενός Bandwidth Broker. Προτείνεται και αξιολογείται ένας προσαρμοστικός αλγόριθμος για αιτήματα κράτησης πόρων που καταφτάνουν νωρίτερα από τον καθορισμένο χρόνο έναρξης της κράτησης. Το γεγονός αυτό επιτρέπει στον αλγόριθμο να συγκεντρώνει ένα σύνολο από πολλαπλά αιτήματα και να κάνει καλύτερη αξιοποίηση του δικτύου, χρησιμοποιώντας την υπάρχουσα βιβλιογραφία για προβλήματα χρονοδρομολόγησης. Η σημασία της παρακολούθησης και της προσαρμογής της υπολογιστικής επιβάρυνσης για τον Bandwidth Broker φαίνεται σαφέστερα όταν υπάρχει μεγάλος ρυθμός άφιξης αιτημάτων, ενώ το ζητούμενο bandwidth για κάθε κράτηση είναι μικρό, όπως στην περίπτωση πολλαπλών VoIP αιτημάτων σε μία σύνδεση υψηλού bandwidth. / The main goal of this dissertation is the study of two of the main developments related to the Internet network layer: the provisioning of Quality of Service guarantees to part of the total traffic traversing ΙΡ networks, as well as the need for upgrading the IPv4 protocol to IPv6. Also goal of this dissertation is the development of mechanisms and algorithms for the effective administration of resources, the best possible fairness in distributing the quality of service, and the possibility of cooperation and interoperability between different domains in an automated way (without the need for human intervention in most cases). For this reason, a number of approaches have been proposed related to Bandwidth Brokers. These approaches are studied in this dissertation, while new algorithms and mechanisms are proposed for the improvement of their operation and performance. IPv4 was capable of supporting QoS mechanisms at the network layer using the TOS field (Type Of Service). IPv6 advances and improves on this idea, by supplying two new fields in the standard header, called Traffic Class and Flow Label, which can be used for this purpose. The usage of these fields, as well as the usage of IPv6 is still at an early stage. However, while IPv6 comes to the foreground and becomes mature enough to replace the dominant IPv4, it is especially interesting to investigate the way that IPv6 QoS capabilities are practically going to be exploited. An important parameter for supporting end-to-end QoS mechanisms is the interaction between multiple domains so that the designated traffic is subjected to preferential treatment along the whole path. The negotiation of this interaction clearly has to be as much automated as possible, if such services are to be widely supported. For this reason, RFC 2638 from IETF has defined the Bandwidth Broker entity. According to the RFC definition, it controls the network load by accepting or rejecting requests for specific bandwidth with QoS guarantees. Bandwidth Brokers only need to establish relationships of limited trust with their peers in adjacent domains, unlike schemes that require the setting of flow specifications in routers throughout an end-to-end path. In practical technical terms, the Bandwidth Broker architecture makes it possible to keep state on an administrative domain basis, rather than at every router and the service definitions of Premium and Assured service make it possible to confine per flow state to just the leaf routers. In the framework of this dissertation we have also studied the admission control module of a Bandwidth Broker. An adaptive algorithm for advance resource reservation requests is proposed and evaluated. The algorithm gathers and evaluates multiple requests in order to better utilize the network, using previous work on timescheduling problems. The importance of monitoring and adapting the computational overhead for the Bandwidth Broker is clearly demonstrated for high request arrival rates and small bandwidth requests, such as the case for multiple VoIP requests that use a high bandwidth link.

With the deployment of 3G networks and gradual implementation of wireless networks, seamless handover between these wireless networks is becoming an increasingly desirable. mSCTP (Mobile Stream Control Transmission Protocol) is a new protocol developed from SCTP (Stream Control Transmission Protocol) to provide seamless handover based on IP networks. This thesis studies how to use this new protocol to handle handovers on transport level between CDMA2000 and WLAN networks. A survey of recently proposed and used mobility protocols is presented, comparing three common handover protocols operating on different layers: MIP (mobile IP) for the network layer, mSCTP for the transport layer and SIP (Session Initial Protocol) for the session layer. The results show mSCTP is the future for mobility support. Lastly, I will present a detailed procedure on how to set up handover testbed between CDMA2000 network and 802.11 WLAN based on mSCTP and the results show that the handover performed between these two networks is fast and smooth but it is affected by the signal strength of the CDMA2000.

O crescimento dos serviços de banda-larga em redes de comunicações móveis tem provocado uma demanda por dados cada vez mais rápidos e de qualidade. A tecnologia de redes móveis chamada LTE (Long Term Evolution) ou quarta geração (4G) surgiu com o objetivo de atender esta demanda por acesso sem fio a serviços, como acesso à Internet, jogos online, VoIP e vídeo conferência. O LTE faz parte das especificações do 3GPP releases 8 e 9, operando numa rede totalmente IP, provendo taxas de transmissão superiores a 100 Mbps (DL), 50 Mbps (UL), baixa latência (10 ms) e compatibilidade com as versões anteriores de redes móveis, 2G (GSM/EDGE) e 3G (UMTS/HSPA). O protocolo TCP desenvolvido para operar em redes cabeadas, apresenta baixo desempenho sobre canais sem fio, como redes móveis celulares, devido principalmente às características de desvanecimento seletivo, sombreamento e às altas taxas de erros provenientes da interface aérea. Como todas as perdas são interpretadas como causadas por congestionamento, o desempenho do protocolo é ruim. O objetivo desta dissertação é avaliar o desempenho de vários tipos de protocolo TCP através de simulações, sob a influência de interferência nos canais entre o terminal móvel (UE User Equipment) e um servidor remoto. Para isto utilizou-se o software NS3 (Network Simulator versão 3) e os protocolos TCP Westwood Plus, New Reno, Reno e Tahoe. Os resultados obtidos nos testes mostram que o protocolo TCP Westwood Plus possui um desempenho melhor que os outros. Os protocolos TCP New Reno e Reno tiveram desempenho muito semelhante devido ao modelo de interferência utilizada ter uma distribuição uniforme e, com isso, a possibilidade de perdas de bits consecutivos é baixa em uma mesma janela de transmissão. O TCP Tahoe, como era de se esperar, apresentou o pior desempenho dentre todos, pois o mesmo não possui o mecanismo de fast recovery e sua janela de congestionamento volta sempre para um segmento após o timeout. Observou-se ainda que o atraso tem grande importância no desempenho dos protocolos TCP, mas até do que a largura de banda dos links de acesso e de backbone, uma vez que, no cenário testado, o gargalo estava presente na interface aérea. As simulações com erros na interface aérea, introduzido com o script de fading (desvanecimento) do NS3, mostraram que o modo RLC AM (com reconhecimento) tem um desempenho melhor para aplicações de transferência de arquivos em ambientes ruidosos do que o modo RLC UM sem reconhecimento. / The growth of broadband services in mobile networks has led to a demand for data with faster and better quality transmissions. The mobile network technology called LTE (Long Term Evolution) or fourth generation (4G) came up with the objective of attending this demand for wireless access to services such as Internet access, online games, VoIP and video conferencing. LTE is part of the specifications of 3GPP Releases 8 and 9 operating in all-IP networks and providing transmission rates above 100 Mbps (DL), 50 Mbps (UL), low latency (10 ms) and compatibility with previous versions of mobile networks, 2G (GSM / EDGE) and 3G (UMTS / HSPA). The TCP protocol designed to operate in wired networks presents poor performance over wireless channels such as mobile cellular networks, due mainly to the characteristics of selective fading, shadowing and high error rates coming from the air interface. As all losses are interpreted as caused by congestion the protocol performance is bad. The objective of this dissertation is to evaluate the performance of several types of the TCP protocols through simulations, under the influence of channel interference between the mobile terminal (UE - User Equipment) and a remote server. For this, the NS3 (Network Simulator version 3) software and the protocols TCP Westwood Plus, New Reno, Reno and Tahoe were used. Results have shown that the TCP Westwood Plus protocol has a better performance than others. The New Reno and Reno TCP protocols had similar performance due to the proposed interference model, which has a uniform distribution and so the possibility of loss of consecutive bits is low on the same transmission window. TCP Tahoe, as expected has shown the worst performance among all because it does not have the fast recovery mechanism and its congestion window keeps coming back to one segment after a timeout. It was also observed that the delay has a greater importance in the performance of TCP when comparing with the bandwidth of the access and backbone links importance, once in the tested scenario the bottleneck was present in the air interface. The simulation performed with noise in the Air Interface, introduced by the NS3 fading script, showed that the RLC AM (acknowledged mode) had a better performance than the RLM UM (Unacknowledged mode).

O crescimento dos serviços de banda-larga em redes de comunicações móveis tem provocado uma demanda por dados cada vez mais rápidos e de qualidade. A tecnologia de redes móveis chamada LTE (Long Term Evolution) ou quarta geração (4G) surgiu com o objetivo de atender esta demanda por acesso sem fio a serviços, como acesso à Internet, jogos online, VoIP e vídeo conferência. O LTE faz parte das especificações do 3GPP releases 8 e 9, operando numa rede totalmente IP, provendo taxas de transmissão superiores a 100 Mbps (DL), 50 Mbps (UL), baixa latência (10 ms) e compatibilidade com as versões anteriores de redes móveis, 2G (GSM/EDGE) e 3G (UMTS/HSPA). O protocolo TCP desenvolvido para operar em redes cabeadas, apresenta baixo desempenho sobre canais sem fio, como redes móveis celulares, devido principalmente às características de desvanecimento seletivo, sombreamento e às altas taxas de erros provenientes da interface aérea. Como todas as perdas são interpretadas como causadas por congestionamento, o desempenho do protocolo é ruim. O objetivo desta dissertação é avaliar o desempenho de vários tipos de protocolo TCP através de simulações, sob a influência de interferência nos canais entre o terminal móvel (UE User Equipment) e um servidor remoto. Para isto utilizou-se o software NS3 (Network Simulator versão 3) e os protocolos TCP Westwood Plus, New Reno, Reno e Tahoe. Os resultados obtidos nos testes mostram que o protocolo TCP Westwood Plus possui um desempenho melhor que os outros. Os protocolos TCP New Reno e Reno tiveram desempenho muito semelhante devido ao modelo de interferência utilizada ter uma distribuição uniforme e, com isso, a possibilidade de perdas de bits consecutivos é baixa em uma mesma janela de transmissão. O TCP Tahoe, como era de se esperar, apresentou o pior desempenho dentre todos, pois o mesmo não possui o mecanismo de fast recovery e sua janela de congestionamento volta sempre para um segmento após o timeout. Observou-se ainda que o atraso tem grande importância no desempenho dos protocolos TCP, mas até do que a largura de banda dos links de acesso e de backbone, uma vez que, no cenário testado, o gargalo estava presente na interface aérea. As simulações com erros na interface aérea, introduzido com o script de fading (desvanecimento) do NS3, mostraram que o modo RLC AM (com reconhecimento) tem um desempenho melhor para aplicações de transferência de arquivos em ambientes ruidosos do que o modo RLC UM sem reconhecimento. / The growth of broadband services in mobile networks has led to a demand for data with faster and better quality transmissions. The mobile network technology called LTE (Long Term Evolution) or fourth generation (4G) came up with the objective of attending this demand for wireless access to services such as Internet access, online games, VoIP and video conferencing. LTE is part of the specifications of 3GPP Releases 8 and 9 operating in all-IP networks and providing transmission rates above 100 Mbps (DL), 50 Mbps (UL), low latency (10 ms) and compatibility with previous versions of mobile networks, 2G (GSM / EDGE) and 3G (UMTS / HSPA). The TCP protocol designed to operate in wired networks presents poor performance over wireless channels such as mobile cellular networks, due mainly to the characteristics of selective fading, shadowing and high error rates coming from the air interface. As all losses are interpreted as caused by congestion the protocol performance is bad. The objective of this dissertation is to evaluate the performance of several types of the TCP protocols through simulations, under the influence of channel interference between the mobile terminal (UE - User Equipment) and a remote server. For this, the NS3 (Network Simulator version 3) software and the protocols TCP Westwood Plus, New Reno, Reno and Tahoe were used. Results have shown that the TCP Westwood Plus protocol has a better performance than others. The New Reno and Reno TCP protocols had similar performance due to the proposed interference model, which has a uniform distribution and so the possibility of loss of consecutive bits is low on the same transmission window. TCP Tahoe, as expected has shown the worst performance among all because it does not have the fast recovery mechanism and its congestion window keeps coming back to one segment after a timeout. It was also observed that the delay has a greater importance in the performance of TCP when comparing with the bandwidth of the access and backbone links importance, once in the tested scenario the bottleneck was present in the air interface. The simulation performed with noise in the Air Interface, introduced by the NS3 fading script, showed that the RLC AM (acknowledged mode) had a better performance than the RLM UM (Unacknowledged mode).

Orientador: Nelson Luis Saldanha da Fonseca / Tese (doutorado) - Universidade Estadual de Campinas, Instituto de Computação / Made available in DSpace on 2018-08-05T04:20:10Z (GMT). No. of bitstreams: 1 Lima_MicheleMaradeAraujoEspindula_D.pdf: 987855 bytes, checksum: 0b48a6acb474d2be5e0d437581cada01 (MD5) Previous issue date: 2005 / Resumo: A ocorrência de congestionamento degrada o desempenho das redes de computadores. Dentre as conseqüências negativas da sua ocorrência cita-se a diminuição da vazão, a perda de pacotes, e o aumento do atraso. Para prevenir e controlar o congestionamento, o protocolo Transmission Control Protocol (TCP) varia a taxa de transmissão de dados de acordo com o nível de congestionamento existente. As políticas de Gerenciamento Ativo de Filas, do Inglês Active Queue Management (AQM), monitoram o nível de ocupação das filas, afim de notificar o congestionamento incipiente aos nós emissores. Esta notificação é realizada através da marca¸c¿ao ou do descarte de pacotes. O sistema de controle de congestionamento em redes TCP/IP, pode ser visto como um sistema de controle por retroalimentação, no qual, a taxa de transmissão dos n'os fontes é ajustada de acordo com o nível de ocupação da fila. Os controladores para o gerenciamento ativo de filas determinam o valor da probabilidade de descarte ou de marcação, buscando a maximização da vazão e a minimiza¸c¿ao das perdas, garantindo, assim, a estabilidade do tamanho da fila independentemente das variações das condições da rede. Nesta tese, são utilizadas técnicas da teoria de controle ótimo para definir uma política ótima de gerenciamento ativo de filas, denominada H2-AQM. A principal característica da H2-AQM é o uso de controladores não racionais, superando-se, assim, a dificuldade de se incorporar no projeto do controlador a garantia de estabilidade em relação ao atraso da retroalimentação. Outrossim, a estabilidade e os objetivos de desempenho do sistema são completamente expressos e solucionados através de desigualdades matriciais lineares, permitindo que os parâmetros do controlador possam ser calculados através da solução de um problema convexo simples. Diferentes controladores operando no mesmo ponto de equilíbrio definem diferentes caminhos entre um ponto qualquer de operação do sistema e o ponto de equilíbrio. Por outro lado, o caminho percorrido para atingir a estabilidade depende dos objetivos usados para projetar o controlador. Nesta tese, é discutida, também, a escolha dos objetivos do projeto de um controlador ótimo para o gerenciamento ativo de filas. Os desempenhos dos diferentes controladores são avaliados e a eficácia do controlador que apresentou o melhor desempenho foi comparado com o desempenho das políticas RED e PI-AQM / Abstract: Congestion is one of the most significant problems in networking. When congestion occurs, the network performance degrades, leading to throughput decrease, delay increase and packet losses. In order to avoid congestion the Transmission Control Protocol (TCP) changes its transmission rate according to the level of congestion. AQM policies notify incipient congestion to TCP source by marking or dropping packets. In TCP/ICP networks, congestion control system can be viewed as a feedback control system in which the transmission rate of the sources are adjusted according to the level of congestion inferred by the queue occupancy. Controllers are responsible for determining the appropriate value of the dropping/marking probability values that stabilizes the queue size regardless of the network condition. In this thesis, optimal control theory is used to conceive an optimal AQM policy, called H2-AQM. The novelty of the proposed approach lies in the use of non-rational controllers that overcomes the difficulty of incorporating guarantees of the stability with respect to the delayed part of the system in the controller design. Furthermore, in the proposed approach stability and performance objectives are completely expressed as Linear Matrix Inequalities (LMIs), thus requiring the solution of a single convex problem for the computation of the controller parameters. Different controllers define different pathes for taking the system state to a target point of equilibrium. Moreover, the path depends on the objectives established for the design of the controller. In this thesis, a discussion on the design of AQM optimal controllers for optimal performance is also presented. The performance produced by different optimal controllers was investigated. The efficacy of the controller which presented the best performance was, then, compared to the performance of both RED and PI-AQM policies / Doutorado / Redes de Computadores / Doutor em Ciência da Computação