Private 5G networks will soon be ubiquitous across the future-generation smart wireless access infrastructures hosting a wide range of performance-critical applications. A high-performing User Plane Function (UPF) in the data plane is critical to achieving such stringent performance goals, as it governs fast packet processing and supports several key control-plane operations. Based on a private 5G prototype imple-mentation and analysis, it is imperative to perform dynamic resource management and orchestration at the UPF. This paper leverages Mobile Edge Cloud-Intelligent Agent (MEC-IA), a logically centralized entity that proactively distributes resources at UPF for various service types, significantly reducing the tail latency experienced by the user requests while maximizing resource utilization. Extending the MEC-IA functionality to MEC layers further incurs data plane latency reduction. Based on our extensive simulations, under skewed uRLLC traffic arrival, the MEC-IA assisted bestfit UPF-MEC scheme reduces the worst-case latency of UE requests by up to 77.8% w.r.t. baseline. Additionally, the system can increase uRLLC connectivity gain by 2.40× while obtaining 40% CapEx savings.

We propose a novel cloud-based data storage solution framework named Data Vaporizer (DV). The proposed framework provides many unique features such as storing data over multiple clouds or storage zones, resistance against organized vendor attacks, maintaining data integrity and confidentiality through client-side processing, fault-tolerance against failure of one or more cloud storage locations and avoids vendor lock-in of data. Data Vaporizer is highly configurable to meet various client data encryption requirements, compliance to industry standards and fault tolerance constraints depending on the nature and sensitivity of the data. To enhance the level of security and reliability, especially to protect data against malicious attacks and secure key management in cloud, DV uses advanced techniques of secret sharing of the keys. The architecture and optimality of data placement and efficient key management algorithm of DV ensure that the solution is highly scalable. The data foot print and subsequent cost incurred by our storage solution is minimal, considering the benefits provided. The initial response for the adoption of DV in actual client scenarios is promising.

A middleware is a piece of software which resides between the main application and the physical infrastructure. Middleware processes the data coming from the physical sensor network and make them available to applications through various API. The conventional middleware available in the literature is very complex to use, because the application developer needs to have knowledge of hardware, embedded software and related interfaces. These middleware are just used for data processing, engineering unit conversion and subsequently data management. In this paper we are describing MOJO, a middleware that converts a physical sensor network into a virtual sensor network. In this virtual sensor networks every node is represented as a Java object. One can develop application using these Java Objects without bothering about the complexity of the physical systems. Our protocol stack is such that any operation done on these Java objects are also reflected back at the real nodes. The changes in the physical nodes are reflected on these virtual nodes. Using this middleware IT developers, not skilled in WSN, can develop application using these Java objects.

In order to meet the increasing demands of content dissemination in Internet, we propose a novel architecture for the future Internet called cache-and-forward (CNF), which transports content as "packages" in a hop-by-hop manner towards the destination, instead of transporting a stream of fragmented packets along an established TCP/IP connection. In this paper, we discuss how the CNF network architecture can be designed for efficient content retrieval. We first introduce several specific services provided in CNF network which are centered around content handling and mobile access. We then give an overview of the CNF protocol stack, which is built on top of IP, and consists of a data plane and a control plane. We provide detailed descriptions of each protocol within both planes. Then we present two caching algorithms, where one involves each CNF router making independent decisions on content caching while the other coordinates node caching within an autonomous system (AS) through hashing. Finally, we gave the initial simulation results to show the performance benefits of hop-by-hop transport and content caching.

In 802.11 wireless links with disabled MAC retries, data and ACK packets within a TCP session collide resulting in packet losses. We show in this paper that in this situation, certain popular optimizations of TCP (fast-recovery) worsen the performance by causing deadlocks that terminate with timeouts. We compare the performance of optimized TCP versions (Reno and NewReno) with an earlier version of TCP (Tahoe) and demonstrate the degradation during fast-recovery. TCP-Tahoe gains 80% in throughput over TCP-Reno and more moderately over TCP-NewReno. A key contribution of this paper is the visualization of TCP dynamics to capture MAC layer collisions between DATA and ACK packets of a TCP session, and the differences in the behavior of protocols in that situation. This case of poor TCP performance due to self-inflicted losses, makes a sound case for decoupling error and flow control algorithms for transport over 802.11 wireless networks.

This paper investigates the TCP simultaneous-send problem (SSP) which arises in infrastructure mode WLANs. In particular it has been observed that for file transfer traffic, 802.11 wireless nodes have a sustained supply of packets to send and hence experience a relatively high rate of MAC contention. For TCP, this results in competition among data and ACK packets for channel access which causes the SSP that deteriorates flow throughput. This SSP can be alleviated by skipping TCP ACKs. Detailed simulation results are presented to demonstrate the usefulness of ACK skipping in various network scenarios such as with MAC retries and multiple TCP flows. The largest improvement is seen for the case of a single TCP flow, and moderate gains are also achieved in cases with multiple streams. For the single TCP stream case with 1 ACK skip and no MAC retries, TCP throughput improves 30% for short-lived and 98% for long-lived TCP transmissions. The paper concludes with potential cross-layer solutions that potentially provide further improvements, including the use of the point coordination function (PCF) to reduce contention between multiple TCP streams and returning ACK packets.

The multimedia broadcast/multicast service (MBMS), which increases network efficiency, and facilitates group-oriented applications, is a service in 3G networks that is about to be standardized. A key security issue for MBMS is the design of a mechanism that controls access to multicast data. Traditionally, access control is done by distributing and maintaining key information. However, the existing architecture of mobile equipment prevents keys from being stored securely on the user side, making the traditional key management insufficient to protect the confidentiality of multicast data. Some discussion in 3GPP has addressed the issue, but the existing scheme achieves the security goal at the cost of high bandwidth overhead, unpredictable peak bandwidth and potential server implosion problems. Currently, no quantified convincing research has been done to study these issues. In this paper, we present new schemes that solve the problems, and quantify their scalability and bandwidth overhead.

Evolving monitoring infrastructure in response to network growth is a critical aspect of network management. Previous work in network management primarily focused on optimizing monitoring systems for static networks. In this paper, we address the problem of optimally upgrading the existing monitoring infrastructure as the network evolves. The problem formulation presented here captures the trade off between adding new monitoring resources vs. disrupting the existing infrastructure. We show that this problem is NP hard and not approximable within a factor better than n/sup /spl epsi// in the general case and no better than log(n) when only shortest paths are considered. We develop a heuristic algorithm and evaluate its performance using simulated network evolution scenarios. We show that in spite of not allowing poller relocation, our adaptive online algorithm has comparable performance to that of the offline algorithm where such constraint does not exist.

The recent reduction in telecommunications spending has increased the importance of network planning to improve the return on investment on the existing network infrastructures. Therefore, tools that help in maximizing the bandwidth efficiency of the network at a minimum cost are essential. Previous work in this area focused on increasing bandwidth efficiency and reliability. In this work, in addition to increasing the bandwidth efficiency, we address the complexity of network management and operations. This issue is explicitly addressed by our novel framework, a simple polynomial time algorithm (SimPol) that achieves optimum network performance (in terms of congestion or bandwidth consumption) using only a small number of paths. The problem formulation is based on splittable multicommodity flows. Using SimPol we show that the total number of paths is at most k+m, where k and m are the numbers of demands and edges in the network, respectively. We extend the basic framework into an integer programming formulation to address the tradeoff between network congestion and the total number of paths. We also use SimPol to address the problem of implementing path/link policies such as bandwidth-limited paths. The performance of SimPol is evaluated through extensive simulations. Using the integer programming approach, we can get exactly one path while losing about 10% to 50% in congestion depending on the number of demands. This congestion is, however, far better than the traditional shortest path routing. The framework is general and can be used in capacity planning for transport networks such as MPLS and ATM.

A virtual private network (VPN) is an overlay network that uses the public network to carry data traffic between corporate sites and users, maintaining privacy through the use of tunnelling protocols and security procedures. In the network-based model, VPN-aware network elements are placed within the network to set up concatenated tunnels between the user/site and enterprise resources to offer intranet VPN and remote access VPN. This paper identifies the important differences between a traditional VPN and the mobile VPN and proposes a hierarchical network architecture to efficiently realize network-based mobile VPNs. We address the problem of optimally provisioning VPN-aware devices, called IP service gateways (IPSGs), in the hierarchical network architecture for mobile VPNs, while taking into account of (1) the cost of links over which VPN tunnels are established, (2) the cost of provisioning a VPN customer on an IPSG, and (3) redundancy in IPSG provisioning for fault tolerance. We develop generic yet powerful problem formulations for different scenarios described above while considering practical requirements of the network elements and business requirements of the VPN service provider. The formulation becomes a set of integer programming problems. We solve several instances of the problem for a few practical cases and discuss their applications in the overall network design.

In a CDMA2000 network, a mobile node (MN) gets packet data service by establishing a PPP session with a packet data serving node (PDSN). The PDSN acts as the muter for the IP packets transported over the PPP session. A packet control function (PCF) sits between the radio access network and the packet network, selects the PDSN for the MN during session setup, and relays the PPP frames between the MN and the PDSN during the session. The PCF selects a PDSN based on the MN's device identity. Due to this, the data services that a mobile user subscribes become tightly coupled with the device that the user registers with the service provider. Because the PCF selects a PDSN based only on mobile device identity, it is unable to select the "best" PDSN to support the services subscribed by a specific user. This work presents the design and implementation of an entity called radio-packet session redirector (RPSR) that works within the current standard and alleviates this shortcoming. The RPSR intercepts a PPP session, parses the user identity, selects a PDSN based on the user identity, creates a PPP session with the selected PDSN and splices the sessions together so that PPP frames can be seamlessly forwarded between the MN and the PDSN with very little overhead. We identify a number of services that RPSR enables in the network. We describe a prototype stand-alone device implementation of the RPSR in the Linux kernel and present performance results.

With the popularity of services like push-to-talk, the need for "always on" services is becoming important for service providers. The paper addresses the problem of supporting always on services in existing and new network architectures. It defines the requirements of always on service, identifies the problems in supporting such a service, and proposes an overlay network based solution to make always on service a reality. Some results from initial prototyping and experimentation are also presented to demonstrate the feasibility of deploying such services.

Designing optimal measurement infrastructure is a key step for network management. In this work we address the problem of optimizing a scalable distributed polling system. The goal of the optimization is to reduce the cost of deployment of the measurement infrastructure by identifying a minimum poller set subject to bandwidth constraints on the individual links. We show that this problem is NP-hard and propose three different heuristics to obtain a solution. We evaluate our heuristics on both hierarchical and flat topologies with different network sizes under different polling bandwidth constraints. We find that the heuristic of choosing the poller that can poll the maximum number of unpolled nodes is the best approach. Our simulation studies show that the results obtained by our best heuristic is close to the lower bound obtained using LP relaxation.

As multicast applications have become widely deployed on the Internet, it is increasingly important to ensure these applications respond to network congestion in a TCP-friendly manner so as to coexist with TCP connections (which constitute the majority of the Internet traffic). We present a RAte-based Congestion COntrOI scheme for Multicast, called RACCOOM, for applications that deploy source-based multicast trees as the communication paradigm. In the absence of packet loss, a RACCOOM session keeps track of the congestion status of the on-tree path with the largest round trip time (called the target path) and adjusts its sending rate using a TCP Vegas [L. S. Brakmo et al., (1995)]-like method. Upon detection of packet loss anywhere in the multicast tree, RACCOOM then responds by reducing its sending rate by half in a TCP-Reno manner. The ACK aggregation method used in RACCOOM prevents ACK implosion and yet provides the sender with a simple but comprehensive view of congestion conditions in the multicast tree. Finally, RACCOOM is equipped with mechanisms to deal with changes of the target path due to traffic change and member join/leave. To achieve TCP-friendliness, we have devised a simple method in RACCOOM to emulate how a TCP connection would behave under the same packet loss and delay characteristics. The results thus derived are used by RACCOOM to online adjust the parameters of its rate adjustment method. Alternatively, we can achieve (weighted) fairness (in terms of bandwidth sharing) among competing RACCOOM connections based on results obtained from feedback control theory. We validate the design, and demonstrate the features, of RACCOOM in ns-2. The encouraging simulation results, coupled with the fact that all the RACCOOM operations except acknowledgment aggregation (which requires modest router support) can be performed at end hosts, suggest that RACCOOM is a practical and yet effective congestion control solution for multicast applications.

Real-time online multi-player games are becoming increasingly popular due to advances in game design and the proliferation of broadband Internet access. However, fairness remains a major challenge when players over large geographic areas participate in a client-server based game together. The paper proposes a game-independent, network-based service, called Sync-MS, that balances the trade-off between response time and fairness. Sync-MS uses two mechanisms, sync-out and sync-in, to address state update fairness and player action fairness, respectively. Two metrics, ahead and behind, measured against the fair order, are defined to-evaluate Sync-MS's fairness performance. Simulation results show that Sync-MS dramatically improves player action fairness for all players while it slightly increases the average response time for players with shorter network delay to the game server.

Most current schemes for multicast routing assume that multicast routers participate both in forwarding multicast packets and in control algorithms for routing, resource reservation, and group management. By separating data and control flow, and by centralizing control in distinct control elements, we have designed a simple and scalable approach to IP multicast that we call Centralized Multicast. We present the details of our approach, a proof of its correctness, analysis of its performance, and a discussion of its advantages over current schemes.

Layered video multicast with retransmissions (LVMR) is a system for distributing video using layered coding over the Internet. The two key contributions of the system are: (1) improving the quality of reception within each layer by retransmitting lost packets given an upper bound on recovery time and applying an adaptive playback point scheme to help achieve more successful retransmission, and (2) adapting to network congestion and heterogeneity using hierarchical rate control mechanism. This paper concentrates on the rate control aspects of LVMR. In contrast to the existing sender-based and receiver-based rate control in which the entire information about network congestion is either available at the sender (in sender-based approach) or replicated at the receivers (in receiver-based approach), the hierarchical rate control mechanism distributes the information between the sender, receivers, and some agents in the network in such a way that each entity maintains only the information relevant to itself. In addition to that, the hierarchical approach enables intelligent decisions to be made in terms of conducting concurrent experiments and choosing one of several possible experiments at any instant of time based on minimal state information at the agents in the network. Protocol details are presented in the paper together with experimental and simulation results to back our claims.

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This paper presents the design, implementation, and performance of a reliable multicast transport protocol (RMTP). The RMTP is based on a hierarchical structure in which receivers are grouped into local regions or domains and in each domain there is a special receiver called a designated receiver (DR) which is responsible for sending acknowledgments periodically to the sender, for processing acknowledgment from receivers in its domain, and for retransmitting lost packets to the corresponding receivers. Since lost packets are recovered by local retransmissions as opposed to retransmissions from the original sender, end-to-end latency is significantly reduced, and the overall throughput is improved as well. Also, since only the DRs send their acknowledgments to the sender, instead of all receivers sending their acknowledgments to the sender, a single acknowledgment is generated per local region, and this prevents acknowledgment implosion. Receivers in RMTP send their acknowledgments to the DRs periodically, thereby simplifying error recovery. In addition, lost packets are recovered by selective repeat retransmissions, leading to improved throughput at the cost of minimal additional buffering at the receivers. This paper also describes the implementation of RMTP and its performance on the Internet.

This paper describes the design and implementation of a multi-cast transport protocol called RMTP. RMTP provides sequenced, lossless delivery of bulk data from one sender to a group of receivers. RMTP achieves reliability by using a packet-based selective repeat retransmission scheme, in which each acknowledgment (ACK) packet carries a sequence number and a bitmap. ACK handling is based on a multi-level hierarchical approach, in which the receivers are grouped into a hierarchy of local regions, with a designated receiver (DR) in each local region. Receivers in each local region periodically send ACKs to their corresponding DR, DRs send ACKs to the higher-level DRs, until the DRs in the highest level send ACKs to the sender, thereby avoiding the ACK-implosion problem. DRs cache received data and respond to retransmission requests of the receivers in their corresponding local regions, thereby decreasing end-to-end latency and improving resource usage. This paper also provides the measurements of RMTP's performance with receivers located at various sites in the Internet.

We present a new approach for conformance testing of protocols specified as a collection of communicating finite state machines (FSMs). Our approach uses a guided random walk procedure. This procedure attempts to cover all transitions in the component FSMs. We also introduce the concept of observers that check some aspect of protocol behavior. We present the result of applying our method to two example protocols: full-duplex alternating bit protocol and the ATM-adaptation-layer-convergence protocol. Applying our procedure to the ATM adaptation layer, 99% of component FSMs edges can be covered in a test with 11692 input steps. Previous approaches cannot do conformance test generation for standard protocols (such as asynchronous transfer mode (ATM) adaptation layer) specified as a collection of communicating FSMs.

Communications networks are traditionally used to bring information together. They can also be used to keep information apart in order to protect personal privacy. A cryptographic protocol specifies a process by which some information is transferred among some users and hidden from others. We show how to implement anonymous credit cards using simple cryptographic protocols. We pose, and solve, a collusion problem which determines whether it is possible for a subset of users to discover information that is designed to be hidden from them during or after execution of the anonymous credit card protocol.

We develop an asymmetric protocol for wireless communication in a step-by-step manner. We start with a very simple protocol and prove its correctness. Then we relax the assumptions of the simple protocol one by one, verifying the correctness of the protocol at each step as we relax the assumptions. This process is continued in a systematic manner until no assumptions are left. The novelty of the paper lies in the way the assumptions are relaxed without violating the correctness properties of the protocol while at the same time making the protocol efficient. The final result is a provably correct protocol which is also efficient for wireless channels.

This paper describes a communications networking technique for funds transfer which combines the privacy of cash transactions with the security, record-keeping and charging mechanisms of credit cards. The scheme uses a communications network and cryptographic protocols to separate information. The company that extends credit to the individual and collects the bill does not have access to the specific purchases, and the shop that sells the merchandise is convinced that it will be paid without learning the individual's identity.

This paper presents the design and analysis of three reliable multicast transport protocols for high speed networks. The novelty of these protocols lies in the technique used in combining the acknowledgments of individual destinations along the underlying multicast tree to prevent acknowledgement implosion and in the technique used in preventing unnecessary retransmission by performing local multicasts. These protocols use the periodic exchange of complete state information between the source and the destinations and a block-based Selective Repeat retransmission scheme to improve the overall performance in a high speed networking environment. Performance of each protocol is analyzed in terms of throughput, end-to-end delay, buffer requirement, acknowledgment traffic and retransmission traffic. Based on this analysis and the complexity of implementation, one of the three protocols is recommended for reliable multicasting in high speed networks.>

Presents and investigates the performance of a secure access scheme to shared information. The primary target application is the electronic newspaper for mobile, wirelessly accessing users. In this application, a dynamically changing set of users is allowed to access the newspaper server. The authors based the solution on the locker key scheme, in which a user's access permission is granted by the server placing a universal encryption key in the user's buffer. The newspaper is then encrypted with the universal key and made public. Some of the salient features of the proposed scheme are: the newspaper is encrypted once and a single copy is stored in the server, the encryption is done off-line, considerably reducing the server congestion, and there is no need to redistribute the universal key upon its change. Furthermore, the authors show that, using some realistic parameter values, the scheme can reduce the access time two to three orders of magnitude over a scheme in which the encryption is performed in real-time on a request-by-request basis.

A theoretical analysis of the fault coverage of conformance test sequences for communication protocols specified as finite state machines is presented. Faults of different types are considered, and their effect on testing is analyzed. The interaction between faults of different categories and the impact it has on conformance testing is investigated. Fault coverage is defined for the testing of both incompletely-specified machines (ISMs) and completely-specified machines (CSMs). An algorithm is presented to generate test sequences with maximal fault coverage for the testing of ISMs. It is then augmented for the testing of CSMs, and finally a technique is presented for generating test sequences which provides guaranteed maximal fault coverage for the conformance testing of communication protocols.>

This paper focuses on a technique to reduce the length of maximal fault coverage test sequences for communication protocols by removing redundant test segments. This approach conceptually begins with all the test segments needed for the generation of maximal fault coverage test sequences, analyzes the structure of the specified finite state machine for the protocol, and shows that certain segments in these tests are unnecessary to guarantee maximal fault coverage. From this analysis an algorithm is proposed for generating the reduced length sequences that still guarantee maximal fault coverage. The authors describe how these tests are in some sense minimal, or near minimal, length test sequences without losing fault coverage.>

An approach for conformance testing of protocols specified as a collection of communicating finite state machines (FSMs) with two parts, pruning and a guided random walk procedure, is presented. First the protocol is pruned to various sets of machines; each set provides only one service. This significantly reduces the test sequence length. Then a guided random walk procedure that attempts to cover all transitions in the component FSMs is used. The results of applying the procedure to the full-duplex alternating bit protocol and the asynchronous transfer mode (ATM) adaptation layer convergence protocol are presented. For the ATM adaptation layer, 99% of component FSMs' edges can be covered in a test with 11692 input steps. Previous approaches cannot generate conformance tests for standard protocols (such as ATM adaptation layer) specified as a collection of communicating FSMs.>

A technique for generating a test sequence for conformance testing of communication protocols is presented. This approach shows that it is possible to generate optimal-length test sequences which include multiple unique input/output (UIO) sequences and overlapping under certain conditions. In the absence of the above-mentioned conditions, a heuristic technique is used to obtain suboptimal solutions which show significant improvement over optimal solutions without overlapping. The technique is illustrated by the example of the NBS Class 4 Transport Protocol (TP4). The computational complexity of the algorithm is compared with that of previous techniques. A brief discussion of bounds on test sequence length is presented, and the results are compared with these bounds.>

A new technique of generating a test sequence for conformance testing of communication protocols is presented. This approach shows that it is possible to generate optimal length test sequences which include multiple unique input output sequences that overlap under certain conditions. In the absence of these conditions, a heuristic technique is used to obtain sub-optimal solutions, which show significant improvement over optimal solutions without overlapping. A comparison is also made of the computational complexity of the author's algorithm with that of existing techniques. A brief discussion of bounds on test sequence length is presented and results are compared against these bounds.>