Higher automation is seen as a key milestone in the evolution of telecom networks. Maintaining a myriad of different technologies over constantly evolving architectures (from 2G-5G) increases operational expenditures and poses real integration and deployment challenges for operators. In addition, the new network dynamism introduced with virtualization, eventually leading to multi-tenancy, will add further complexities for the operator in managing the network. This paper outlines the essential role of open and closed control loops in achieving higher levels of network automation, identifies the related challenges that operators face in deploying control loops in live environments and describes preliminary, standards-oriented solutions.

Supporting end-to-end network slices and services across operators has become an important use case of study for 5G networks as can be seen by 5G use cases published in 3GPP, ETSI as well as NGMN. This paper presents the in- depth architecture, implementation and experiment on a multi-domain orchestration framework that is ab le to deploy such multi-operator service as well as monitor the service for SLA compliance. Our implemented architecture allows operators to abstract their sensitive details while exposing the relevant amount of information to support inter-operator slice creation. Our experiment shows that the implemented framework is capable of creating services across operators while fulfilling the respective service requirements.

Virtual Network Embedding (VNE) is considered a key technology to instantiate and operate Data and Control planes in next generation (5G) SDN-based Networks. Within this domain, Network Resource Management (NRM) is an essential feature to allow efficient resource utilisation, to enable network slicing and to guarantee fairness among the supported QoS classes. This paper presents and evaluates three alternative NRM policies: Full Sharing, Full Split and Russian Dolls. Policies define how different QoS classes share the available bandwidth on per link basis. The policies have been integrated in a MIP-based Virtual Link Mapping formulation (VLM+) supporting multi-constrained end to end QoS. Simulation results show different policies can suit different network operator's requirements. Also, results highlight Russian Dolls significantly outperforms other policies in terms of Embedding Rate and Link Utilisation, still preserving fairness among QoS Classes. VLM+ Convergence Time has also been evaluated, showing all policies are compatible with timing requirements for a real 5G system implementation.

This paper presents VLM+, a Virtual Link (VL) Embedding algorithm supporting Quality of Service (QoS). VLM+ has been developed evolving its Mixed Integer Programming (MIP) based Virtual Link Mapping algorithm precursor (VLM), whose analytical model has been enhanced including the ability to fulfill VL end-to-end QoS requirements and to prioritise VL requests according to their QoS class or revenue profile. Additionally, VLM+ supports three different physical resource sharing policies, this providing Physical Infrastructure Providers (PIPs) with high flexibility in the allocation of resources to embed QoS-constrained VLs. VLM+ has been designed and evaluated targeting next generation networks, conceived around SDN and NFV paradigms and expected to support, among others, delay critical services. VLM+ performance has been evaluated via simulation and compared to VLM, Shortest Path First (SPF) and Constrained SPF algorithms. Promising quantitative results demonstrate VLM+ ability to embed VLs fulfilling end-to-end QoS requirements, still achieving an efficient resource utilisation and at a moderate cost in terms of increased complexity and convergence time.

One of the main goals of future telecom networks is to achieve softwarization of network functions. A requirement to achieve this softwarization is the ability to construct on-demand networks across multi-domain network and/or cloud service providers (NCSP). Most current algorithms and MILP formulations that solve the multi-domain embedding problem work on a flat infrastructure using simplified physical resource models. In this paper we propose an abstraction of the physical network domains to make the multiple domains appear as a pseudo flat infrastructure enabling re-use of existing flat infrastructure embedding algorithms with a few modifications. We incorporate these modifications in our previously proposed embedding ILP formulation. Our results show that we can speed up the process of embedding in large multi-domain networks considerably while trading off some efficiency in resource utilization due to abstraction.

Virtual network embedding is considered an important problem to solve in order to make infrastructure virtual-ization economically reasonable. The most efficient algorithms proposed so far define link and node mappings as optimal solutions of Integer Programming (IP) problems. They exhibit reasonably good performance only for small problem instance sizes, including few tens of nodes and links per physical substrate, few nodes and links per virtual request and a dozen of virtual requests to handle in parallel. However, we find these instances too small to be of any practical use. To address this scalability issue, we propose VNetMapper, an algorithm to solve the virtual network embedding problem based on an integer program formulation with appropriately selected objective function, variables and the set of constraints tuned to give an optimal performance. We show through simulations that VNetMapper can quickly, within seconds, solve large problem instances, involving physical substrates with hundreds of nodes and thousands of links and batches of hundreds of virtual network requests. By identifying exact properties that make VNetMapper so fast and scalable, we present guidelines for designing scalable integer programs.

The development of methodologies to manage and orchestrate virtualised resources and network functions is a fundamental enabler for optimally utilising physical ICT infrastructures. Algorithms for optimal location (embedding) of network functions, IT and CT resources, services and corresponding states, especially at the network edge, will enable new business models and provide a key competitive advantage to network administrators. This paper introduces a novel Mixed Integer Programming (MIP) formulation for a coordinated node and link mapping onto the underlying network infrastructure. Extensive simulation results show that the proposed algorithm outperforms prior art formulations: two digit gains were attained in terms of resources utilisation, embedding, revenues and, especially convergence time. The proposed methodology is applicable to a number of relevant use cases, as constraints and objective functions can be flexibly defined by network operators.

High performance systems require high levels of reliability. Many functions involved in telecommunication and IT networks have reliability requirements that can only be achieved by introducing redundant resources. In the telecom sector, recently, there has been a significant effort on moving carrier grade systems and functions to virtualized network infrastructure. The management and coordination of those virtualized systems to achieve an optimal mapping (or embedding) to the physical resources that host them is known as virtual resource orchestration. In our prior work we introduced a novel model, based on Mixed Integer Programming (MIP) problem formulation, as one significant way of achieving this optimality in embedding. This paper extends the model to include reliability requirements, improving prior art techniques as well as implementing a novel approach, denoted as reliability assurance. The confidence of the target reliability of the embedded virtual graphs can be traded with the efficiency of the substrate utilization. Extensive simulation results show that our model provides embedding rates and infrastructure utilization comparable with prior art while fulfilling high reliability requirements.

Reducing operational costs and increasing energy efficiency are critical for mobile operators. One accepted way to tackle these issues is to make use of virtualization to share the same physical infrastructure among several virtual networks. However, sharing physical infrastructure presents a number of challenges, such as security issues, conflicting hardware control, and fair resource sharing. In our earlier work, we analyzed these challenges and demonstrated how virtualized mobile operator networks can address them. However, to create and manage virtual mobile networks, an automated configuration system is required. This system must incorporate the diverse nature of technology, ownership, and geographic domains into an overarching single virtual network. This paper discusses the main challenges in creating such an architecture. We then present our proposed architecture and describe how it meets these challenges. To conclude, we present our experiences in the real-world implementation of this architecture.

Mounting capital and operational expenditure and reducing revenue per bit is straining the resources of mobile communication network operators. The future mobile communication networks will have to be dynamic; to adapt and re-adapt to the prevailing network conditions to increase operational efficiency. This paper presents the various use cases for such a Reconfigurable Mobile Network (RMN). We believe that the best way of currently achieving this reconfigurability is via virtualization techniques. Virtualization has been touted as the main solution that will ease the revolution to the future Internet and possibly result in a greener and more efficient operation of the current Internet. This paper investigates whether the concept of network virtualization can be re-applied to a mobile operator network to create RMN. We present detailed use cases for a reconfigurable mobile network and discuss the derived requirements on the RMN infrastructure. We see that a number of requirements, such as absolute isolation, may be technically infeasible in the current end-to-end mobile network.

The concept of CloudNets, virtual networks connecting cloud resources, has recently attracted much interest from both academic as well as business sides. CloudNets can realize the vision of affordable customized infrastructures. In particular, such networks are expected to be offered even in federated environments with multiple providers. Inter-provider communication about requirements or provisioning of truly customized virtual environments however require a powerful flexible resource description language (RDL). While extensibility and expressiveness seem to be natural requirements for such a language, we identify another less intuitive requirement affecting all actors (or stakeholders) in their economic benefits: the possibility to omit arbitrary specification details and to remainvaguewhile at the same time describing real world scenarios. Not only may a description language ignoring this constraint easily become too bulky to use, it is also likely to force players to focus on details they are not interested in or lack the knowledge to map their actual requirements to. This paper identifies detailed requirements for an RDL to allow for topology and requirement communication in business scenarios. Furthermore, we present the FleRD flexible resource description language for multi-provider virtual network architectures. FleRD is fully incorporated in our own CloudNet prototype architecture.

Synchronization between locations is an important factor for enabling remote shared experiences. Still, experimental data on what is the acceptable synchronization level is scarce. This paper discusses the synchronization requirements for watching online videos together - a popular set of services that recreate the shared experience of watching TV together by offering tools to communicate while watching. It studies the noticeability and annoyance of synchronization differences of the video being watched, as well as the impact on users' feelings of togetherness, both for voice chat and text chat. Results of an experiment with 36 participants show that when using voice chat, users notice synchronization differences sooner, are more annoyed and feel more together than when using text chat. However, users with high text chat activity notice synchronization differences similar to participants using voice chat.

This paper presents Eureka: a generic methodology of measuring the instantaneous (per second) bandwidth usage of networked games and applications in run time. Eureka starts with constructing a priority queue and sending low priority traffic through it. Then, the application under study is started with its packets marked as high priority. These packets therefore displace the previously existing low priority packets on the priority queue. We can measure this displacement and thus, the bandwidth used by the application. Unlike tcpdump this can be done while the application is running and without knowing any application specific data. In this paper we use Eureka to study the bandwidth usage of Massive Multiplayer Online Games (MMOGs), BZFlag and World of Warcraft-Trial, along with the 3D multi-site collaborative environments. A major strength of our methodology is that it provides run-time estimations of the bandwidth usage without interfering with the application, the environment, or the server/peer data. Moreover, Eureka is independent of the application and new infrastructure or tools are not required.

This paper discusses the architectural challenges of transitioning from services to experiences. In particular, it focuses on evolution from traditional IPTV services to more social scenarios, in which groups of people in different locations watch synchronized multimedia content together. In addition to the multimedia content, the shared experiences envisioned in this article provide a real-time communication channel between the participants. Based on an implemented architecture, this paper identifies a number of challenges and analyze them. The most important challenges highlighted in this article include: shared experience modeling, universal session handling, synchronization, and quality of service. This article is the first stone paving the way for a truly interoperable ecosystem, which can offer cross-domain experiences to the users.

This paper presents a new scheduling algorithm for real time network delivery of packets over Diffserv networks for delay sensitive applications. We name the networks that support this algorithm as Estimated Service (Estserv) networks. These networks, for real time packets, estimate the probability of the packet meeting its deadline and schedule it according to this estimation. This paper validates, given this estimation mechanism, the better performance of the scheduling algorithm over traditional solutions. We show that using Estserv for delay sensitive applications, we can provide out-of-band scheduling, save bandwidth on packets with expired deadlines and handle bursts without loosing the scalability of Diffserv. We show with the help of an implementation in the Linux kernel's ip-forwarding part, that, given the estimation value, Estserv performs better than Diffserv in terms of deadlines, while still saving bandwidth.

This paper demonstrates the advantages of using structured multimedia documents for session management and media distribution in ubiquitous environments. We show how document manipulations can be used to perform powerful operations such as content to context adaptation and presentation continuity. When consuming media in ubiquitous environments, where the set of devices surrounding a user may change, dynamic media adaptation and session transfer become primary requirements. This paper presents a working system, based on a representative scenario, in which multimedia content is distributed and adapted to a movable user to best suit his/her contextual situation. The implemented scenario includes the following scenes: content selection using a personal mobile phone, content distribution to the most suitable device according to the user's context, and presentation continuity when the user moves to another location. This paper introduces the underlying document manipulations that turn the scenario into a working system.

This thesis aims to provide a better scheduling algorithm for Real-Time delivery of packets. A number of emerging applications such as VoIP, Tele-immersive environments, distributed media viewing and distributed gaming require real-time delivery of packets. Currently the scheduling algorithms used to decide the priority ordering over packets only consider the deadline as the parameter. This may not be optimal, since it ignores the network traffic over the routes that these packets may take. The idea of this thesis is to propose a method of calculating a probability measure of each packet meeting it's deadline at every intermediate node. This probability measure is based on the time left for deadline to expire, the number of nodes further in the packets route to reach it's destination and the traffic on this route. The algorithm assumes a certain level on time synchronisation over the network.

This paper presents an approach for media presentation continuity in playback mode. We use the term presentation continuity over session transfer since our solution is at the presentation layer. Previous research on this topic has focused on transferring a particular stream or set of related streams at the sessions layer for live broadcasting or conferencing sessions. We argue that in the realm of service oriented architectures, such as telecom operator networks, this approach does not take full advantage of the particular case of media playback. Our mechanism presents an alternative to the traditional approach, which i) Lowers network control plane overhead, thus reducing chances of presentation consistency loss ii) Lowers network data overhead due to lesser need for transcoding iii) Delegates presentation consistency issues, such as inter-media synchronisation, to the media player iv) Dynamically adapts the presentation to the new target devices without transcoding. Finally, we present experimental results which show that our approach is implementable in acceptable time bounds.

This demo presents a new approach for media presentation continuity in playback mode. We use the term presentation continuity over session transfer since our solution is at the presentation layer. Previous research on this topic has focused on transferring a particular stream or set of related streams at the sessions layer. Our approach presents an alternative, recognising the fact that a user is connected to a media presentation, which, may be composed of multiple sessions. The advantages of our approach are i) Lower network control plane overhead, thus reducing chances of semantic presentation loss ii) Lower network data overhead due to lesser need for transcoding iii) delegating presentation semantic issues, such as inter-media synchronisation, to the player iv) dynamically adapt the presentation to the new target devices without transcoding.