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

Scaling up VoIP: Transport Protocols and Controlling Unwanted Communication Requests

Millions of people worldwide use voice over IP (VoIP) services not only as cost-effective alternatives to long distance and international calls but also as unified communication tools, such as video conferencing. Owing to the low cost of new user accounts, each person can easily obtain multiple accounts for various purposes. Rich VoIP functions combined with the low cost of new accounts and connections attract many people, resulting in a dramatic increase in the number of active user accounts. Internet telephony service providers (ITSPs), therefore, need to deploy VoIP systems to accommodate this growing demand for VoIP user accounts. Attracted people also include bad actors who make calls that are unwanted to callees. Once ITSPs openly connect with each other, unwanted bulk calls will be at least as serious a problem as email spam. This dissertation studies how we can reduce load both on ITSPs and end users to ensure continuing the success of VoIP services. From ITSPs' perspective, the scalability of VoIP servers is of importance and concern. Scalability depends on server implementation and the transport protocol for SIP, VoIP signaling. We conduct experiments to understand the impact of connection-oriented transport protocols, namely, TCP and SCTP, because of the additional costs of handling connections. Contradicting the negative perception of connection-oriented transport protocols, our experimental results demonstrate that the TCP implementation in Linux can maintain comparable capacity to UDP, which is a lightweight connection-less transport protocol. The use of SCTP, on the other hand, requires improving the Linux implementation since the not-well-tested implementation makes a server less scalable. We establish the maximum number of concurrent TCP or SCTP connections as baseline data and suggest better server configurations to minimize the negative impact of handling a large number of connections. Thus, our experimental analysis will also contribute to the design of other servers with a very large number of TCP or SCTP connections. From the perspective of end users, controlling unwanted calls is vital to preserving the VoIP service utility and value. Prior work on preventing unwanted email or calls has mainly focused on detecting unwanted communication requests, leaving many messages or calls unlabeled since false positives during filtering are unacceptable. Unlike prior work, we explore approaches to identifying a "good" call based on signaling messages rather than content. This is because content-based filtering cannot prevent call spam from disturbing callees since a ringing tone interrupts them before content is sent. Our first approach uses "cross-media relations.'' Calls are unlikely to be unwanted if two parties have been previously communicated with each other through other communication means. Specifically, we propose two mechanisms using cross-media relations. For the first mechanism, a potential caller offers her contact addresses which might be used in future calls to the callee. For the second mechanism, a callee provides a potential caller with weak secret for future use. When the caller makes a call, she conveys the information to be identified as someone the callee contacted before through other means. Our prototype illustrates how these mechanisms work in web-then-call and email-then-call scenarios. In addition, our user study of received email messages, calls, SMS messages demonstrates the potential effectiveness of this idea. Another approach uses caller's attributes, such as organizational affiliation, in the case where two parties have had no prior contact. We introduce a lightweight mechanism for validating user attributes with privacy-awareness and moderate security. Unlike existing mechanisms of asserting user attributes, we design to allow the caller to claim her attributes to callees without needing to prove her identity or her public key. To strike the proper balance between the ease of service deployment and security, our proposed mechanism relies on transitive trust, through an attribute validation server, established over transport layer security. This mechanism uses an attribute reference ID, which limits the lifetime and restricts relying parties. Our prototype demonstrates the simplicity of our concept and the possibility of practical use

Scaling up VoIP: Transport Protocols and Controlling Unwanted Communication Requests

Millions of people worldwide use voice over IP (VoIP) services not only as cost-effective alternatives to long distance and international calls but also as unified communication tools, such as video conferencing. Owing to the low cost of new user accounts, each person can easily obtain multiple accounts for various purposes. Rich VoIP functions combined with the low cost of new accounts and connections attract many people, resulting in a dramatic increase in the number of active user accounts. Internet telephony service providers (ITSPs), therefore, need to deploy VoIP systems to accommodate this growing demand for VoIP user accounts. Attracted people also include bad actors who make calls that are unwanted to callees. Once ITSPs openly connect with each other, unwanted bulk calls will be at least as serious a problem as email spam. This dissertation studies how we can reduce load both on ITSPs and end users to ensure continuing the success of VoIP services. From ITSPs' perspective, the scalability of VoIP servers is of importance and concern. Scalability depends on server implementation and the transport protocol for SIP, VoIP signaling. We conduct experiments to understand the impact of connection-oriented transport protocols, namely, TCP and SCTP, because of the additional costs of handling connections. Contradicting the negative perception of connection-oriented transport protocols, our experimental results demonstrate that the TCP implementation in Linux can maintain comparable capacity to UDP, which is a lightweight connection-less transport protocol. The use of SCTP, on the other hand, requires improving the Linux implementation since the not-well-tested implementation makes a server less scalable. We establish the maximum number of concurrent TCP or SCTP connections as baseline data and suggest better server configurations to minimize the negative impact of handling a large number of connections. Thus, our experimental analysis will also contribute to the design of other servers with a very large number of TCP or SCTP connections. From the perspective of end users, controlling unwanted calls is vital to preserving the VoIP service utility and value. Prior work on preventing unwanted email or calls has mainly focused on detecting unwanted communication requests, leaving many messages or calls unlabeled since false positives during filtering are unacceptable. Unlike prior work, we explore approaches to identifying a "good" call based on signaling messages rather than content. This is because content-based filtering cannot prevent call spam from disturbing callees since a ringing tone interrupts them before content is sent. Our first approach uses "cross-media relations.'' Calls are unlikely to be unwanted if two parties have been previously communicated with each other through other communication means. Specifically, we propose two mechanisms using cross-media relations. For the first mechanism, a potential caller offers her contact addresses which might be used in future calls to the callee. For the second mechanism, a callee provides a potential caller with weak secret for future use. When the caller makes a call, she conveys the information to be identified as someone the callee contacted before through other means. Our prototype illustrates how these mechanisms work in web-then-call and email-then-call scenarios. In addition, our user study of received email messages, calls, SMS messages demonstrates the potential effectiveness of this idea. Another approach uses caller's attributes, such as organizational affiliation, in the case where two parties have had no prior contact. We introduce a lightweight mechanism for validating user attributes with privacy-awareness and moderate security. Unlike existing mechanisms of asserting user attributes, we design to allow the caller to claim her attributes to callees without needing to prove her identity or her public key. To strike the proper balance between the ease of service deployment and security, our proposed mechanism relies on transitive trust, through an attribute validation server, established over transport layer security. This mechanism uses an attribute reference ID, which limits the lifetime and restricts relying parties. Our prototype demonstrates the simplicity of our concept and the possibility of practical use

User generated content for IMS-based IPTV

Includes abstract.Includes bibliographical references.Web 2.0 services have been on the rise due to improved bandwidth availability. Users can now connect to the internet with a variety of portable devices which are capable of performing multiple tasks. Due to this, services such as Voice over IP (VoIP), presence, social networks, instant messaging (IM) and Internet Protocol television (IPTV) to mention but a few, started to emerge...This thesis proposed a framework that will offer user-generated content on an IMS-Based IPTV and the framework will include a personalised advertising system..

Linking session based services with transport plane resources in IP multimedia subsystems.

The massive success and proliferation of Internet technologies has forced network operators to recognise the benefits of an IP-based communications framework. The IP Multimedia Subsystem (IMS) has been proposed as a candidate technology to provide a non-disruptive strategy in the move to all-IP and to facilitate the true convergence of data and real-time multimedia services. Despite the obvious advantages of creating a controlled environment for deploying IP services, and hence increasing the value of the telco bundle, there are several challenges that face IMS deployment. The most critical is that posed by the widespread proliferation ofWeb 2.0 services. This environment is not seen as robust enough to be used by network operators for revenue generating services. However IMS operators will need to justify charging for services that are typically available free of charge in the Internet space. Reliability and guaranteed transport of multimedia services by the efficient management of resources will be critical to differentiate IMS services. This thesis investigates resource management within the IMS framework. The standardisation of NGN/IMS resource management frameworks has been fragmented, resulting in weak functional and interface specifications. To facilitate more coherent, focused research and address interoperability concerns that could hamper deployment, a Common Policy and Charging Control (PCC) architecture is presented that defines a set of generic terms and functional elements. A review of related literature and standardisation reveals severe shortcomings regarding vertical and horizontal coordination of resources in the IMS framework. The deployment of new services should not require QoS standardisation or network upgrade, though in the current architecture advanced multimedia services are not catered for. It has been found that end-to-end QoS mechanisms in the Common PCC framework are elementary. To address these challenges and assist network operators when formulating their iii NGN strategies, this thesis proposes an application driven policy control architecture that incorporates end-user and service requirements into the QoS negotiation procedure. This architecture facilitates full interaction between service control and resource control planes, and between application developers and the policies that govern resource control. Furthermore, a novel, session based end-to-end policy control architecture is proposed to support inter-domain coordination across IMS domains. This architecture uses SIP inherent routing information to discover the routes traversed by the signalling and the associated routes traversed by the media. This mechanism effectively allows applications to issue resource requests from their home domain and enable end-to-end QoS connectivity across all traversed transport segments. Standard interfaces are used and transport plane overhaul is not necessary for this functionality. The Common PCC, application driven and session based end-to-end architectures are implemented in a standards compliant and entirely open source practical testbed. This demonstrates proof of concept and provides a platform for performance evaluations. It has been found that while there is a cost in delay and traffic overhead when implementing the complete architecture, this cost falls within established criteria and will have an acceptable effect on end-user experience. The open nature of the practical testbed ensures that all evaluations are fully reproducible and provides a convenient point of departure for future work. While it is important to leave room for flexibility and vendor innovation, it is critical that the harmonisation of NGN/IMS resource management frameworks takes place and that the architectures proposed in this thesis be further developed and integrated into the single set of specifications. The alternative is general interoperability issues that could render end-to-end QoS provisioning for advanced multimedia services almost impossible

Autonomic Overload Management For Large-Scale Virtualized Network Functions

The explosion of data traffic in telecommunication networks has been impressive in the last few years. To keep up with the high demand and staying profitable, Telcos are embracing the Network Function Virtualization (NFV) paradigm by shifting from hardware network appliances to software virtual network functions, which are expected to support extremely large scale architectures, providing both high performance and high reliability. The main objective of this dissertation is to provide frameworks and techniques to enable proper overload detection and mitigation for the emerging virtualized software-based network services. The thesis contribution is threefold. First, it proposes a novel approach to quickly detect performance anomalies in complex and large-scale VNF services. Second, it presents NFV-Throttle, an autonomic overload control framework to protect NFV services from overload within a short period of time, allowing to preserve the QoS of traffic flows admitted by network services in response to both traffic spikes (up to 10x the available capacity) and capacity reduction due to infrastructure problems (such as CPU contention). Third, it proposes DRACO, to manage overload problems arising in novel large-scale multi-tier applications, such as complex stateful network functions in which the state is spread across modern key-value stores to achieve both scalability and performance. DRACO performs a fine-grained admission control, by tuning the amount and type of traffic according to datastore node dependencies among the tiers (which are dynamically discovered at run-time), and to the current capacity of individual nodes, in order to mitigate overloads and preventing hot-spots. This thesis presents the implementation details and an extensive experimental evaluation for all the above overload management solutions, by means of a virtualized IP Multimedia Subsystem (IMS), which provides modern multimedia services for Telco operators, such as Videoconferencing and VoLTE, and which is one of the top use-cases of the NFV technology

Mobile Networks

The growth in the use of mobile networks has come mainly with the third generation systems and voice traffic. With the current third generation and the arrival of the 4G, the number of mobile users in the world will exceed the number of landlines users. Audio and video streaming have had a significant increase, parallel to the requirements of bandwidth and quality of service demanded by those applications. Mobile networks require that the applications and protocols that have worked successfully in fixed networks can be used with the same level of quality in mobile scenarios. Until the third generation of mobile networks, the need to ensure reliable handovers was still an important issue. On the eve of a new generation of access networks (4G) and increased connectivity between networks of different characteristics commonly called hybrid (satellite, ad-hoc, sensors, wired, WIMAX, LAN, etc.), it is necessary to transfer mechanisms of mobility to future generations of networks. In order to achieve this, it is essential to carry out a comprehensive evaluation of the performance of current protocols and the diverse topologies to suit the new mobility conditions