Integrated Support for Handoff Management and Context-Awareness in Heterogeneous Wireless Networks

The overwhelming success of mobile devices and wireless communications is stressing the need for the development of mobility-aware services. Device mobility requires services adapting their behavior to sudden context changes and being aware of handoffs, which introduce unpredictable delays and intermittent discontinuities. Heterogeneity of wireless technologies (Wi-Fi, Bluetooth, 3G) complicates the situation, since a different treatment of context-awareness and handoffs is required for each solution. This paper presents a middleware architecture designed to ease mobility-aware service development. The architecture hides technology-specific mechanisms and offers a set of facilities for context awareness and handoff management. The architecture prototype works with Bluetooth and Wi-Fi, which today represent two of the most widespread wireless technologies. In addition, the paper discusses motivations and design details in the challenging context of mobile multimedia streaming applications

A novel middleware for the mobility management over the Internet

The features of mobility, which enormously impact on how communication is evolving into the future, represent a particular challenge in today’s wireless networking research. After an identification and evaluation of the gap between the discontinuities of the communication service inherent to the physical layer of mobile networks and the continuity requirements issue from the stream centric multimedia applications, we propose a novel middleware 3MOI (Middleware for the Mobility Management Over the Internet) which can perform efficient and context-aware mobility management and satisfy new mobility requirements such as dynamical location management, fast handover, and continuous connection support

Support infrastructures for multimedia services with guaranteed continuity and QoS

Advances in wireless networking and content delivery systems are enabling new challenging provisioning scenarios where a growing number of users access multimedia services, e.g., audio/video streaming, while moving among different points of attachment to the Internet, possibly with different connectivity technologies, e.g., Wi-Fi, Bluetooth, and cellular 3G. That calls for novel middlewares capable of dynamically personalizing service provisioning to the characteristics of client environments, in particular to discontinuities in wireless resource availability due to handoffs. This dissertation proposes a novel middleware solution, called MUM, that performs effective and context-aware handoff management to transparently avoid service interruptions during both horizontal and vertical handoffs. To achieve the goal, MUM exploits the full visibility of wireless connections available in client localities and their handoff implementations (handoff awareness), of service quality requirements and handoff-related quality degradations (QoS awareness), and of network topology and resources available in current/future localities (location awareness). The design and implementation of the all main MUM components along with extensive on the field trials of the realized middleware architecture confirmed the validity of the proposed full context-aware handoff management approach. In particular, the reported experimental results demonstrate that MUM can effectively maintain service continuity for a wide range of different multimedia services by exploiting handoff prediction mechanisms, adaptive buffering and pre-fetching techniques, and proactive re-addressing/re-binding

Multimedia session continuity in the IP multimedia subsystem : investigation and testbed implementation

Includes bibliographical references (leaves 91-94).The advent of Internet Protocol (IP) based rich multimedia services and applications has seen rapid growth and adoption in recent years, with an equally increasing user base. Voice over IP (VoIP) and IP Television (IPTV) are key examples of services that are blurring the lines between traditional stove-pipe approach network infrastructures. In these, each service required a different network technology to be provisioned, and could only be accessed through a specific end user equipment (UE) technology. The move towards an all-IP core network infrastructure and the proliferation of multi-capability multi-interface user devices has spurred a convergence trend characterized by access to services and applications through any network, any device and anywhere

A solution for transparent mobility with route optimization in the IP multimedia subsystem

This paper presents TRIM+, an architecture for transparent mobility management with route optimization in IMS based networks. The design of our architecture is based on a previous work referred to as TRIM. TRIM was originally devised to provide transparent mobility support in the IMS, although transparency came at the cost of using a suboptimal data path between communicating end points. TRIM+ maintains transparency as a design criterium, and thus end-user applications, running at the mobile node and its correspondent communication peers, are unaware of mobility management procedures. Additionally, the proposed design defines a set of route optimization procedures, allowing compliant devices to use the optimal data path for media communications. Furthermore, TRIM+ addresses packet loss management in scenarios where the media path cannot be maintained during the handover of the MN. To this end, our architecture enables the MN to request buffering capacity in its home network to temporarily store incoming media traffic during the handover, which would otherwise be dropped. This mechanism, as well as route optimization procedures, are executed transparently to the end-user applications running at the communicating end points. As a proof-of-concept, we have implemented a software prototype of the TRIM+ architecture, deploying it over a real IMS testbed. By means of a set of experiments, we have validated the mechanisms proposed in this paper, considering both UDP and TCP user traffic.This article has been partially granted by the Madrid Commu nity through the MEDIANET project (S 2009/TIC 1468), and by the European Community through the CROWD project (FP7 ICT 318115). The work of Ignacio Soto has been partially sup ported through the I MOVING project (TEC2010 18907).Publicad

Session Continuity in Heterogeneous Networks: A SIP-based Proactive Handover Scheme

Today, the computation power and storage capability on mobile devices are increasing rapidly, and together with new interactive services this creates a demand for more bandwidth. To keep track of this evolution, the use of heterogeneous networks has gained focus. Instead of only using one type of network, it is desired to utilize all carriers available on a device, and hence choose the one best suited. This thesis describes and discusses different approaches to session continuity. The different solutions include Mobile IP, Generic Access Networks (GAN), and proposals based on the Session Initiation Protocol (SIP). Current solutions are not satisfactory in the terms of handover delay, and hence best suited for non-realtime applications. Furthermore it is proposed an application-layer handover scheme for session continuity in heterogeneous networks, which will significantly reduce the handover time. This handover scheme has been implemented, and tests show that the handover time is significantly reduced. Keywords: Session Continuity, Handover, Heterogeneous Networks, SIP, VoIP, GAN

QoS monitoring in real-time streaming overlays based on lock-free data structures

AbstractPeer-to-peer streaming is a well-known technology for the large-scale distribution of real-time audio/video contents. Delay requirements are very strict in interactive real-time scenarios (such as synchronous distance learning), where playback lag should be of the order of seconds. Playback continuity is another key aspect in these cases: in presence of peer churning and network congestion, a peer-to-peer overlay should quickly rearrange connections among receiving nodes to avoid freezing phenomena that may compromise audio/video understanding. For this reason, we designed a QoS monitoring algorithm that quickly detects broken or congested links: each receiving node is able to independently decide whether it should switch to a secondary sending node, called "fallback node". The architecture takes advantage of a multithreaded design based on lock-free data structures, which improve the performance by avoiding synchronization among threads. We will show the good responsiveness of the proposed approach on machines with different computational capabilities: measured times prove both departures of nodes and QoS degradations are promptly detected and clients can quickly restore a stream reception. According to PSNR and SSIM, two well-known full-reference video quality metrics, QoE remains acceptable on receiving nodes of our resilient overlay also in presence of swap procedures

Provider-Controlled Bandwidth Management for HTTP-based Video Delivery

Over the past few years, a revolution in video delivery technology has taken place as mobile viewers and over-the-top (OTT) distribution paradigms have significantly changed the landscape of video delivery services. For decades, high quality video was only available in the home via linear television or physical media. Though Web-based services brought video to desktop and laptop computers, the dominance of proprietary delivery protocols and codecs inhibited research efforts. The recent emergence of HTTP adaptive streaming protocols has prompted a re-evaluation of legacy video delivery paradigms and introduced new questions as to the scalability and manageability of OTT video delivery. This dissertation addresses the question of how to enable for content and network service providers the ability to monitor and manage large numbers of HTTP adaptive streaming clients in an OTT environment. Our early work focused on demonstrating the viability of server-side pacing schemes to produce an HTTP-based streaming server. We also investigated the ability of client-side pacing schemes to work with both commodity HTTP servers and our HTTP streaming server. Continuing our client-side pacing research, we developed our own client-side data proxy architecture which was implemented on a variety of mobile devices and operating systems. We used the portable client architecture as a platform for investigating different rate adaptation schemes and algorithms. We then concentrated on evaluating the network impact of multiple adaptive bitrate clients competing for limited network resources, and developing schemes for enforcing fair access to network resources. The main contribution of this dissertation is the definition of segment-level client and network techniques for enforcing class of service (CoS) differentiation between OTT HTTP adaptive streaming clients. We developed a segment-level network proxy architecture which works transparently with adaptive bitrate clients through the use of segment replacement. We also defined a segment-level rate adaptation algorithm which uses download aborts to enforce CoS differentiation across distributed independent clients. The segment-level abstraction more accurately models application-network interactions and highlights the difference between segment-level and packet-level time scales. Our segment-level CoS enforcement techniques provide a foundation for creating scalable managed OTT video delivery services

Sip Based Mobile Voice Over Ip Client For Wireess Networks

Tez (Yüksek Lisans) -- İstanbul Teknik Üniversitesi, Fen Bilimleri Enstitüsü, 2008Thesis (M.Sc.) -- İstanbul Technical University, Institute of Science and Technology, 2008Bu tez SIP tabanlı mobile bir VoIP istemcisinin tasarımını ve gerçeklenmesini tanımlar. Bu tez temelde çoktürel ağlar üzerinde çalışabilen bir VoIP istemcisi tasarımının çözülmesi gereken iki sorununun üzerinde yogunlaşır. Birinci ve en zorlu sorun farklı erişim teknolojileri arasında kullanıcıya fark ettirmeden yer değişim desteği sağlanmasıdır. Bu tezde, kullanıcıya fark ettirmeden el değiştirme yönetimi, uygulama katmanında, multimedya oturumunu başlatmak, sonlandırmak ve değiştirmek için kullanılan Oturum Başlatma Protokolü (SIP) kullanılarak ele alınmıştır. SIP yaygın bir şekilde kabul edilmekte olan bir VoIP standartıdır. Kullanıcıya fark ettirmeden el değiştirmeyi destekleyebilmek için, VoIP istemcisi üzerinde çalışan SIP tabanlı bir bağlantı yöneticisi önerilmiştir. Bağlantı yöneticisi yeni ağlar keşfettiğinde, adaylar listesinden bir ağ seçer ve hali hazırda yürütülmekte olan iletişimi kullanıcıya fark ettirmeden yeni ağ arayüzüne aktarır. Dolayısı ile, bu birim Wi-Fi, 3G gibi çoktürel ağlar arasında dolaşmayı sağlar. İkinci sorun ise, en kaliteli çağrı (arama) desteğini sağlamaktır. En kaliteli çağrı desteği, iletişim kurmak isteyen tarafların farklı türden ağlara bağlı olmaları durumunda, VoIP uygulamasının iletişim tipine (yarı-çift yönlü yada tam-çift yönlü) karar vermebilmesi demektir. Örneğin, eğer iletişim kurmak isteyen taraflardan biri bir GSM ağındaysa, en iyi çağrı kalitesini yakalayabilmek için, iletişim yarı-çift yönlü olarak kurulmalıdır. Bu tez, bahsedilen özelliği desteklemek için, istemci tabanlı bir karar mekanizması önerir. Bu karar mekanizması, iletişim kurulmak istenen tarafa, istemcinin içinde bulunduğu ağa göre belirlenmiş iletişim tipini içeren bir davet iletisi gönderir. Diğer istemci bu davet iletisini aldıktan sonra, aynı karar mekanizması, iletişimi “bas-konuş VoIP” yada “tam-çift yönlü VoIP” olarak ayarlar.This thesis describes the design and the implementation of a SIP-based mobile VoIP client. It mainly focuses on two challenges of designing a VoIP client which works on heterogeneous network environments. One and the most challenging problem is the provision of seamless mobility support among different access technologies. In this thesis, seamless handover management is handled at the application layer by using Session initiation protocol (SIP), which is used to initiate, terminate, and modify multimedia session. SIP is becoming a widely accepted standard for VoIP. To support seamless handover, a SIP based connection manager is proposed on VoIP client application. As new networks are discovered by the connection manager, it selects a new network from the candidate list and transfers the current communication to the new network interface seamlessly. Therefore, this module provides roaming across heterogeneous networks such as Wi-Fi, 3G. Second problem is providing the best effort call quality support. It means that if the communication parties are in dissimilar networks, the VoIP application should decide the communication type (half-duplex or full-duplex). For instance, if one of the communication parties is in a GSM network, then the communication should be established as a half-duplex manner to achieve best call quality. This thesis proposes a client-based decision mechanism to support this property. This decision mechanism sends an invite message including the communication type (half-duplex or full-duplex) of the client according to the network in which it operates to the other communication party. After the other client receives this invite message, same decision mechanism adjusts the communication as either a “push to talk VoIP” or a “full-duplex VoIP”.Yüksek LisansM.Sc

Network-based IP flow mobility support in 3GPPs evolved packet core

Includes bibliographical references.Mobile data traffic in cellular networks has increased tremendously in the last few years. Due to the costs associated with licensed spectrum, Mobile Network Operators (MNOs) are battling to manage these increased traffic growths. Offloading mobile data traffic to alternative low cost access networks like Wi-Fi has been proposed as a candidate solution to enable MNOs to alleviate congestion from the cellular networks. This dissertation investigates an offloading technique called IP flow mobility within the 3rd Generation Partnership Project (3GPP) all-IP mobile core network, the Evolved Packet Core (EPC). IP flow mobility would enable offloading a subset of the mobile user‟s traffic to an alternative access network while allowing the rest of the end-user‟s traffic to be kept in the cellular access; this way, traffic with stringent quality of service requirements like Voice over Internet Protocol (VoIP) would not experience service disruption or interruption when offloaded. This technique is different from previous offloading techniques where all the end-user‟s traffic is offloaded. IP flow mobility functionality can be realised with either host- or network-based mobility protocols. The recommended IP flow mobility standard of 3GPP is based on the host-based mobility solution, Dual-Stack Mobile IPv6. However, host-based mobility solutions have drawbacks like long handover latencies and produce signaling overhead in the radio access networks, which could be less appealing to MNOs. Network-based mobility solutions, compared to the host-based mobility solutions, have reduced handover latencies with no signaling overhead occurring in the radio access network. Proxy Mobile IPv6 is a networkbased mobility protocol adapted by 3GPP for mobility in the EPC. However, the standardisation of the Proxy Mobile IPv6-based IP flow mobility functionality is still ongoing within 3GPP. A review of related literature and standardisation efforts reveals shortcomings with the Proxy Mobile IPv6 mobility protocol in supporting IP flow mobility. Proxy Mobile IPv6 does not have a mechanism that would ensure session continuity during IP flow handoffs or a mechanism enabling controlling of the forwarding path of a particular IP flow i.e., specifying the access network for the IP flow. The latter mechanism is referred to as IP flow information management and flow-based routing. These mechanisms represent the basis for enabling the IP flow mobility functionality. To address the shortcomings of Proxy Mobile IPv6, this dissertation proposes vi enhancements to the protocol procedures to enable the two mechanisms for IP flow mobility functionality. The proposed enhancements for the session continuity mechanism draw on work in related literature and the proposed enhancements for the IP flow information management and flow-based routing mechanism are based on the concepts used in the Dual- Stack Mobile IPv6 IP flow mobility functionality. Together the two mechanisms allow the end-user to issue requests on what access network a particular IP flow should be routed, and ensure that the IP flows are moved to the particular access network without session discontinuity