Transmission Control Protocol (TCP) is the dominant transport protocol in the Internet and supports many of the most popular Internet applications, such as the World Wide Web (WWW), file transfer and e-mail. TCP congestion control algorithms dynamically learn the network bandwidth and delay characteristics of a network and adapt its performance to changes in traffic so as to avoid network collapse. TCP is designed to perform well in traditional wireline networks with the assumptions that packet losses are mainly due to network congestion and random bit error rate (BER) is negligible. However, networks with wireless links suffer from significant packet losses due to random bit errors and handoffs. Hence TCP performs poorly in networks with wireless links because it treats any packet loss in the network to be a result of network congestion and slows down its transmission rate, or even cause the TCP sender to experience unnecessary timeouts, further reducing its performance. The development of advance wireless networks, such as WiFi, UMTS and WiMAX, make it necessary to find ways to improve TCP's efficiency and resource utilization, as well as improve the user's experience and reduce latency times. In order to find effective solutions to this effect, packet losses across wireless links should be distinguished from congestion related packet losses. In this thesis, we concentrate on two main strategies for enabling the TCP congestion control mechanism to determine the cause for a packet loss. One is a proxy-based mechanism that monitors the radio network interface and sends radio network feedback (RNF) to the TCP sender with the status of the wireless link. The other one is an end-to-end mechanism, in which the packet error pattern is used as the system metric to fine-tune the congestion control mechanism. It also presents an analytical model of TCP with enhanced recovery mechanism for wireless environments. In a proxy-based mechanism, TCP sender is explicitly informed of any effects caused by wireless links. However, the implementation technique is network dependent. We have proposed and developed three proxy-based schemes; the radio network feedback (RNF) scheme over an 802.11 WLAN network, the radio network controller (RNC) feedback over a UMTS network and a wireless enhancement proxy (WENP) over both the 802.11 WLAN and UMTS networks. The RNF scheme is introduced at the 802.11 WLAN base station that monitors the TCP packet flows over the wireless links, detects wireless packet losses and provides feedback to the TCP sender using one of the TCP header reserved control bits, called RNF flag. TCP Reno is modified to utilize the radio network feedback to distinguish the losses due to wireless effects form the congestion and fine-tuned to perform wireless enhanced fast retransmit and fast recovery mechanisms. The RNF scheme is implemented using the OPNET tool, and the simulation results show that the TCP performance is significantly improved. The RNC feedback mechanism, similar to the RNF scheme, is developed and implemented in a UMTS network. The GPRS Tunneling Protocol (GTP) layer of the UMTS Radio Network Control (RNC) protocol stack was modified to detect and notify the TCP sender of the wireless packet losses, which is the main difference between the RNF and RNC mechanisms. The simulation results shows that the RNC feedback mechanism significantly improves the TCP performance compared to that of standard TCP over UMTS. The wireless enhancement proxy (WENP) is developed to minimize spurious TCP timeouts over wireless networks and implemented in both 802.11 WLAN and UMTS networks. WENP extends the proposed RNF and RNC feedback mechanisms to detect both wireless packet losses and large delays across the wireless link, and to notify the TCP sender of these events with the aid of two reserved bits in the TCP header. TCP Reno is further modified to utilize the WENP feedback to distinguish both wireless packet losses from congestion losses and spurious timeouts from normal timeouts. It is also fine-tuned to perform both the wireless enhanced fast retransmit and fast recovery mechanism and the timeout mechanism. The simulation results demonstrate that the proposed scheme markedly improves the TCP performance compared to that of standard WLAN and UMTS implementations. An end-to-end early packet loss recovery (EPLR) mechanism that modifies the TCP Reno fast retransmit algorithm to detect packet losses early and to speed up the packet recovery process to reduce the number of TCP timeouts over networks with heavy packet losses, such as wireless networks is also presented. TCP Reno with EPLR scheme is implemented in a UMTS network and its performance is compared with that of TCP Reno and New Reno. Simulation results shows that Reno with EPLR improves the TCP performance and application response time significantly compared to that of both Reno and New Reno by reducing the TCP timeouts, which is the main cause of degradation of the TCP performance in a wireless environment. Finally, we develop an analytical TCP throughput model with enhanced TCP Reno fast retransmit algorithm to avoid timeouts. The model captures the TCP fast retransmit mechanism and expresses the steady state congestion window and throughput as a function of network utilization factor, round trip time (RTT) and loss rate. Another new feature added to the model is dynamic adjustment of the congestion window size depending on the packet drop rates. This speeds up the packet recovery process and reduces the number of TCP timeouts over networks with heavy packet losses. The proposed model is implemented over a UMTS network and its performance is compared with that of TCP Reno. Simulation results show that the proposed model reduces the TCP timeouts and improves the TCP performance compared to that of TCP Reno. It is also found that the model provides a very good match to the steady-state congestion window behavior.

Due to terminal mobility and change of service area, efficient IP mobility support is an important aspect in UMTS networks in order to provide mobile users negligible packet loss rate and low handover latency, and thus some level of guaranteed quality-ofservice (QoS) to support real-time applications. 3G/UMTS has been specified and implemented as an end-to-end mobile communications system. The underlying WCDMA access systems manage radio access handover (layer 1) and provide linklayer mobility (layer 2) in terms of connection setup and resource management. For the UMTS nodes to have seamless connectivity with the Internet, the UMTS core networks need to be able to support continuous and no network service session handover (layer 3 and above). A long IP handover latency results in high packet loss rate and severely degrades its end-to-end transport level performance. Network-layer handover latency has therefore been regarded as one of the fundamental limitations in IP-based UMTS networks. Therefore, it is crucial to provide efficient network-layer mobility management in UMTS/WCDMA networks for seamless end-to-end TCP connection with the global Internet. Mobility of UMTS nodes necessitates extra functionalities such as user location tracking, address registration and handover related mechanisms. The challenge to provide seamless mobility in UMTS requires localised location management and efficient IP handover management. Mobile IPv6 protocol offers a better mobility support as the extended IPv6 features with mobility mechanism are integrated to the mobile nodes. To mitigate the effect of lengthy IP handover latency, two well-known handover reducing mechanisms based on Mobile IPv6 support have been proposed in the literature. They are designed with hierarchical network management and address pre-configuration mechanism. Hierarchical management aims to reduce the network registration time, and fast-handover attempts to minimise the address resolution delay. S-MIP (Seamless Mobile IP) integrates the key benefits of the above IP mobility mechanisms coupled with local retransmission scheme to achieve packet lossless and extremely low handover latency, operating in WLAN environments. In this thesis, we explore the possible Mobile IP solutions and various IP handover optimisation schemes in IPv6 to provide seamless mobility in UMTS with the global Internet. It aims at developing an optimised handover scheme that encompasses the packet lossless and extremely low handover latency scheme in S-MIP, and applying it into the UMTS/WCDMA packet data domain. Therefore, the hybrid UMTS-SMIP architecture is able to meet the requirements of delay sensitive real-time applications requiring strict delay bound, packet lossless and low handover latency performance for end-to-end TCP connection during a UMTS IP-based handover. The overall seamless handover architecture in UMTS facilitates integrated, scalable and flexible global IP handover solution enabling new services, assuring service quality and meeting the user???s expectations in future all-IP UMTS deployment. The viability of the seamless mobility scheme in UMTS is reflected through and validated in our design model, network protocol implementation, and service architecture. We illustrate the performance gained in QoS parameters, as a result of converged UMTS-SMIP framework compared to other Mobile IPv6 variants. The simulation results show such a viable and promising seamless handover scheme in UMTS on IP handover latency reduction on its end-to-end TCP connection.

Zugl.: Aachen, Techn. Hochsch., Diss., 2006.

Techn. University, Diss., 2005--Darmstadt.

Techn. Univ., Diss.--Braunschweig, 2005.

Zugl.: Berlin, Techn. University, Diss., 2008.

University, Diss., 2003--Dortmund.

Η ΕΕΕ αποτελείται από τέσσερα κεφάλαια και δύο παραρτήματα, ενώ για την καλύτερη κατανόηση και εξοικείωση του αναγνώστη με τις έννοιες και όρους που χρησιμοποιούνται στη βιομηχανία των κινητών επικοινωνιών παρατίθεται ένα σχετικό γλωσσάριο στο τέλος του κειμένου. Το πρώτο κεφάλαιο έχει τίτλο “εισαγωγή στα κυψελωτά συστήματα κινητής τηλεφωνίας” και αποτελεί ουσιαστικά μια εισαγωγή στις βασικές αρχές της κινητής κιυψελωτής τηλεφωνίας. Η εισαγωγή αυτή είναι απαραίτητη προκειμένου να γίνει ομαλά η μετάβαση σε σύνθετες έννοιες που διέπουν την λειτουργία των συστημάτων κινητής τηλεφωνίας. Το δεύτερο κεφάλαιο έχει τον τίτλο “το κανάλι μετάδοσης στις κινητές ραδιοεπικοινωνίες” και αναφέρεται στις ιδιότητες και τα χαρακτηριστικά που λαμβάνουν χώρα κατά την ασύρματη μετάδοση της πληροφορίας. Οι ιδιότητες των ηλ/κων κυμάτων, και της ατμόσφαιρας που αποτελεί το μέσο μετάδοσης στις ασύρματες επικοινωνίες καθώς και η μορφολογία του περιβάλλοντος ραδιομετάδοσης είναι μερικές από τις παραμέτρους που διαδραματίζουν καθοριστικό ρόλο στην ποιότητα των κινητών επικοινωνιών και για τον λόγο αυτό μελετώνται και εν συνεχεία μοντελοποιούνται προκειμένου να επιτύχουμε την μεγαλύτερη δυνατή απόδοση των συστημάτων αυτών. Το τρίτο κεφάλαιο είναι αφιερωμένο αποκλειστικά στο παγκόσμιο σύστημα κινητής τηλεφωνίας G.S.M. Ως γνωστόν, το G.S.M είναι το πιο διαδεδομένο σύστημα κινητών επικοινωνιών και ως εκ τούτου μία εργασία που αναφέρεται σε συστήματα κινητής τηλεφωνίας έχει ως σημείο αναφοράς το G.S.M. Το τέταρτο και τελευταίο κεφάλαιο της ΕΕΕ αναφέρεται στο καθολικό σύστημα κινητών επικοινωνιών UMTS. Το UMTS είναι το σύστημα κινητών επικοινωνιών 3ης γενιάς, το οποίο ουσιαστικά έκανε πραγματικότητα την πάγια απαίτηση από την πλευρά των χρηστών – συνδρομητών για επικοινωνία “οπουδήποτε, οποτεδήποτε, οτιδήποτε”. Η εργασία κλείνει με δύο παραρτήματα. Το παράρτημα Α αποτελεί μία εισαγωγική περιγραφή στο σύστημα σηματοδοσίας Νο 7. Στις περισσότερες περιπτώσεις ο χρήστης κινητής τηλεφωνίας αγνοεί την ύπαρξη του δικτύου σηματοδοσίας το οποίο ουσιαστικά αλλά αφανώς και αθόρυβα «τρέχει» παράλληλα με το δίκτυο μετάδοσης πληροφοριών. Δίνεται λοιπόν η ευκαιρία μέσα από το παράρτημα αυτό να γίνει γνωστή η ύπαρξη του δικτύου σηματοδοσίας Νο 7 (SS#7), όπου περιγράφονται τα λειτουργικά χαρακτηριστικά του (διαστρωματωμένη αρχιτεκτονική, σημεία σηματοδοσίας κλπ.) ενώ γίνεται αναφορά στα είδη των μηνυμάτων σηματοδοσίας και στον τρόπο που ανταλλάσσονται τα μηνύματα αυτά μεταξύ των δομικών στοιχείων του δικτύου SS#7. Στο παράρτημα Β αναφέρονται κάποια βασικά στοιχεία από την «δημοφιλή» τεχνική κωδικοποίησης Linear Predictive Coding (LPC), η οποία στην Ελληνική αποδίδεται ως Γραμμική Προγνωστική Κωδικοποίηση. Η LPC είναι η τεχνική εκείνη που χρησιμοποιείται στο σύνολο σχεδόν των ψηφιακών συστημάτων επικοινωνίας, χάρη στις εγγενείς ιδιότητές της, και επομένως βρίσκει επίσης εφαρμογή και στο G.S.M, ως ένα κατεξοχήν ψηφιακό σύστημα κινητής τηλεφωνίας. Ωστόσο, η εφαρμογή της LPC δεν περιορίζεται μόνο στο G.S.M αλλά εκτείνεται μέχρι τα σημερινά συστήματα 3ης γενιάς. Τέλος, θα ήθελα να αναφέρω δύο λόγια για τις δυσκολίες που συνάντησα καθόλη την διάρκεια συγγραφής της εργασίας αυτής. Η ΕΕΕ, με την παρούσα μορφή, είναι το αποτέλεσμα μιας πολύμηνης προσπάθειας και συνεχούς μελέτης ενός ιδιαίτερα σύνθετου και πολυδιάστατου αντικειμένου έρευνας, όπως είναι τα συστήματα κινητής τηλεφωνίας. Παρά το γεγονός ότι η βιβλιογραφία υπήρξε επαρκής, οι έννοιες και οι τεχνικές δυσκολίες με τις οποίες ήρθα, πολλές φορές, αντιμέτωπος ήταν κάτι το πρωτόγνωρο, το οποίο αρκετές φορές μου έδινε την εντύπωση του ανυπέρβλητου. Θα ήθελα λοιπόν, να ζητήσω εκ των προτέρων την κατανόηση του αναγνώστη για οποιαδήποτε λάθη ή παραλείψεις οι οποίες φυσικά με βαρύνουν στο ακέραιο. / -

In this thesis we investigate, implement and evaluate a solution for introducing the TransmissionControl Protocol (TCP) into the software of a load generator. The load generator is a simulator usedfor simulating end-user generated activities in the Universal Mobile Telecommunication System(UMTS) network. The purpose of simulating traffic on the network is in this case to verify thefunctionality and robustness of the Radio Network Controller (RNC) node within the UMTSnetwork.TCP is a protocol that provides reliable data transfer over unreliable underlying networkprotocols. It is used as the main transport protocol of the Internet, thus it is also used in the UMTSnetwork in order to provide connectivity for user equipment, such as 3G mobile phones, to servicesover the Internet. For the load generator to be able to produce more realistic traffic scenarios is itdesirable to give it the ability to use TCP.This thesis presents a solution of the problem where an open-source implementation of the TCPfunctionality was chosen, ported to the running platform of the load generator and then tested in asimulated test environment. The choice of the open source implementation of TCP was made byperforming an investigation of available options. In the investigation an open source TCP/IP stackcalled lwIP was chosen. lwIP was then ported to the running platform of the load generator bywrapping and modify the source code. The tests of the ported TCP implementation were made in asimulated test environment with focus on testing basic TCP functionality. The tests showed that theTCP implementation produced provided the basic functionality that was asked for.

A local call admission control (CAC) algorithm for third generation wireless networks was designed and implemented, which allows for the simulation of network throughput for different spreading factors and various mobility scenarios. A global CAC algorithm is also implemented and used as a benchmark since it is inherently optimized; it yields the best possible performance but has an intensive computational complexity. Optimized local CAC algorithm achieves similar performance as global CAC algorithm at a fraction of the computational cost. Design of a dynamic channel assignment algorithm for IEEE 802.11 wireless systems is also presented. Channels are assigned dynamically depending on the minimal interference generated by the neighboring access points on a reference access point. Analysis of dynamic channel assignment algorithm shows an improvement by a factor of 4 over the default settings of having all access points use the same channel, resulting significantly higher network throughput.