Energy consumption in Information and Communication Technology (ICT) is 10% of the total energy consumed in industrial countries. Recently, Virtualized Network Environment (VNE) has been emerged in this technology. Therefore, it is essential to develop novel techniques that reduce VNE's energy consumption. In this paper, we formulate a Binary Integer Linear Program (BILP) that reconfigures already allocated virtual networks to minimize VNE's link power consumption, during off-peak periods. Because the formulated BILP is NP-hard, a novel heuristic algorithm is also suggested. The simulation results confirm the proposed solutions save notable amount of energy in VNE's substrate links, during off-peak hours.

By simultaneously providing live video and audio contents to millions of users around the world, peer-to-peer live video streaming (P2P LVS) has become one of the most popular Internet applications in recent years. However, current P2P LVS software has problems such as non-smooth playback and long start-up delay for end users. To address these issues, we design a P2P-based multi-bit Trie structure, called Bottom-Up Trie (BU-Trie), for distributing P2P live contents. Different from other approaches, BU-Trie is a Trie formed and built inversely from leaf nodes (or child nodes) back to the root node (or parent node). This architecture consists of two phases: a diffusion phase and a swarming phase. The main design goal of the diffusion phase is to group the local peers together by discovering physical locations of peers, and design the paths for fast distributing live streams from the source node to end users. The objective of the swarming phase is to find an optimal way for exchanging the video stream chunks within a local group. We propose an algorithm called Most Popular Chunk First (MPCF) and apply it for the swarming phase for efficient chunk exchange. Performance evaluation of the proposed BU-Trie shows that, when compared to other approaches, the sequential throughput of video chunks is increased. The inter-domain traffic, the traffic between different Internet service providers (ISPs), is reduced as well. Such a reduction would benefit carriers economically.

In this work, we investigate the optimal dynamic packet scheduling policy in a wireless relay network (WRN). We model this network by two sets of parallel queues, that represent the subscriber stations (SS) and the relay stations (RS), with random link connectivity. An optimal policy minimizes, in stochastic ordering sense, a cost function of the SS and RS queue sizes. We prove that, in a system with symmetrical connectivity and arrival distributions, a policy that tries to balance the lengths of all the system queues, at every time slot, is optimal. We use stochastic dominance and coupling arguments in our proof. We also provide a low-overhead algorithm for optimal policy implementation.

With rapidly growing deployment, SIP has become a main signaling protocol for IP telephony and multimedia sessions in the Internet. SIP employs a retransmission mechanism to maintain its reliability. Recent server collapse due to emergency-induced call volume in carrier networks indicates that message retransmissions triggered by various SIP timers make the overload worse. The built-in overload control mechanism cannot handle overload conditions effectively. Since the retransmissions caused by the overload introduce more overhead rather than reliability, we suggest mitigating the overload by reducing the retransmission rate. We propose a novel algorithm to detect the potential overload at the downstream servers and control retransmission message rate from upstream servers to mitigate the overload at the downstream servers. We investigate two typical overload scenarios caused by demand burst and server slow down respectively. OPNET simulations demonstrate that (1) the proposed solution can help the overloaded downstream server to cancel its overload effectively after it resumes its normal operation status; (2) without the overload control algorithm applied, the overload at the downstream server may propagate or migrate to its upstream servers.

Recent collapses of SIP servers in the real carrier networks indicate that the built-in SIP overload control mechanism cannot mitigate overload effectively. In this paper, we investigate the root cause of SIP server crash by studying the impact of the retransmission on the queuing delay of the overloaded server. The transient overload may introduce the excessive queuing delay, thus triggering unnecessary retransmissions to exacerbate the overload. Therefore, we adopt a control-theoretic approach that models the overloaded downstream server and its upstream server as a feedback control system. Then we design a PI rate controller to restrict the retransmission rate based on the queuing delay. We derive the guidelines for choosing PI controller gains to ensure the system stability. Our OPNET simulation results demonstrate that our proposed control theoretic approach can mitigate the SIP overload effectively, thus preventing the SIP network collapse.

We present an algorithm to implement “most balancing” packet scheduling policies in a discrete-time multi-server system of parallel queues with independent random queue-server connectivity. The MB policies are characterized by minimizing the total difference in queue lengths at every time slot. The proposed algorithm produces the server allocation strategy that achieves the minimum “imbalance index” at every time slot. This algorithm has a reduced computational complexity compared to a full-search through the set of all feasible policies. We also provide a low-complexity approximate implementation algorithm that performs close to the exact algorithm. Simulation results confirm our claims.

Retransmission mechanism helps SIP maintain its reliability, but it can also make an overload worse. Recent server collapses due to emergency-induced call volume in carrier networks indicate that the built-in overload control mechanism cannot handle overload conditions effectively. Since the retransmissions caused by the overload are redundant, we suggest mitigating the overload by controlling redundant message ratio to an acceptable level. Using control-theoretic approach, we model the interaction of an overloaded downstream server with its upstream server as a feedback control system. Then we develop an adaptive PI control algorithm to mitigate the overload at the downstream server by controlling the retransmission message rate of its upstream servers. By performing OPNET simulations on two typical overload scenarios, we demonstrate that: (1) without overload control algorithm applied, the overload at the downstream server may propagate to its upstream servers; (2) our control-theoretic solution not only mitigate the overload effectively, but also achieve a satisfactory target redundant message ratio.

The presence service provides event-based notifications of the status of contacts. Currently duplicate notifications must be sent directly to a set of watchers in unicast messages, putting unneeded strain on outgoing network links of the source node. In this article, we propose an extension to the Session Initiation Protocol (SIP) to distribute notifications through a tree of watchers using network coordinates to build the tree. Our simulation results show traffic costs halved by using the extension without significantly adding delay.

As a main signaling protocol for multimedia sessions in the Internet, SIP (Session Initiation Protocol) introduces a retransmission mechanism to maintain the reliability for its realtime transmission. However, retransmission will make the server overload worse. Recent collapse of SIP servers due to emergencyinduced call volume indicates that the built-in SIP overload control mechanism cannot prevent the server from overload collapse under heavy load. In this paper, we apply a MMPP (Markov-Modulated Poisson Process) model to analyze the queuing mechanism of SIP server under two typical service states. The MMPP model allows us to investigate the probability of SIP retransmissions. By performing numerous experiments statistically to verify SIP retransmission probability calculated by MMPP model, we find that high retransmission probability caused by short demand surge or reduced server processing capacity during maintenance period may overload and crash a server. We run simulations using time-series directly to observe and analyze the system performance of an overloaded SIP server. This is much faster than event-driven simulation. Numerical results demonstrate that low resource utilization corresponds to low retransmission probability. However, a utilization as low as 20% cannot always guarantee a SIP system stability upon a temporal server slowdown or a short period of demand burst.

The IP Multimedia Subsystem (IMS) is envisioned as the next-generation IP-based multimedia system that integrates data, speech, and video network services over both wireless and wireline networks. In this paper, we focus on optimizing the cost of SIP server deployment in an IMS network. To reflect the traffic loads on the servers, a flow-based model is used to characterize the SIP traffic. Formulated as a linear programming problem, the cost optimization involves mapping a logical IMS core network topology into a physical network topology. Three potential mapping strategies are proposed. Each strategy's specific constraints are incorporated into the mathematical formulation of the problem. A numerical example of each strategy is presented, and the discussion on the formulations is provided.

In this paper, the use of a centralized server to assist cognitive radio users in accessing bands in licensed spectrums is proposed. Typical cognitive radios are opportunistic users of spectrum bands. Therefore, they must scan the spectrum to detect existing users in order to avoid interference. The use of a centralized server can remove the need for spectrum scanning if all users inform the server about their presence. The server will coordinate and distribute channels to cognitive radio users using auctioning mechanisms. Our approach removes the need for cognitive radio users to spectrum scan. Scanning can be costly in terms of time and power consumption. In addition, collisions between users due to hidden node problem can be removed. The use of a centralized server allows for higher layer solution that would allow users of different wireless technologies to communicate. Due to its flexibility of use across different wireless networks, SIP is adopted as the communication protocol between the central server and the primary and secondary users of the licensed spectrum. Using a SIP server to coordinate channel allocation through auctioning approach can generate revenue for the incumbent network. Results show that revenue can be generated while still meeting the goal of efficient spectrum utilization.

The IP Multimedia Subsystem (IMS) defined by the 3rdGeneration Partnership Project (3GPP) and 3GPP2 provides a platform for the provision of multimedia services with quality of service (QoS). In addition, this service architecture allows third-party vendors to create advanced multimedia and multisession services across wireless and wireline network access. The Session Initiation Protocol (SIP) supports the signaling and session management functions of these services; therefore, the SIP performance is critical to the services' quality of experience. Thus, in order to conduct an SIP performance evaluation, an efficient yet representative model for SIP signaling traffic is needed. In this article, we provide an in-depth flow analysis of a number of SIP session procedures defined in IMS and quantify the SIP signaling traffic at flow level. By utilizing the signaling flow analysis, the workload of servers can be predicted with a simple mathematical calculation. The complex correlation structure of the workloads across different signaling servers is naturally captured by the flow concept we introduced. This model also allows for flexibility when expanding the SIP session procedures in IMS networks. According to the simulations that we carried out using OPNET, the model we proposed is proven to be acceptable.

IP Multimedia Subsystem (IMS) provides a common control layer over which the services can be easily accessed by fixed and mobile users. In order to guarantee performance to both service providers and end service subscribers, careful management and maintenance within the control layer must be fulfilled. In this paper, we propose a solution that utilizes a controlling entity to track the utilization of hotspot Session Initiation Protocol (SIP) server (i.e., S-CSCF) and to initiate de-registration procedure so that a sufficient number of subscribers at the overloaded S-CSCF are re-associated with the other S-CSCF(s) for subsequent session services. The proposed solution redistributes and smoothes the future traffic load over individual S-CSCF in an automatic manner. The goodness of the proposal is assessed through simulations using OPNET.

Effective and efficient management of wireless network resources is attracting more and more research attention, due to the rapid growing deployment of wireless mesh and ad hoc networks and to the increasing demand for Quality of Service (QoS) support in these networks. This paper proposes an adaptive network resource management scheme in the popular IEEE 802.11 random access MAC by adaptively adjusting the minimum contention window sizes of traffic flows. First, a novel generalized processor sharing (GPS) model is presented for the IEEE 802.11 random access MAC revealing the relationship between the minimum contention window size of a traffic flow and the amount of network resource this flow can receive. Using this GPS MAC model, a feedback control system model for the proposed adaptive network resource management system is developed, by directly extending our previous work in wireline GPS networks. Based on the feedback control system model, adaptive P and adaptive PI controllers are designed, and their performances are studied in simulations. Simulation results show that by using the designed controllers, the proposed adaptive network resource management approach is able to provide guaranteed distinct QoS support to traffic flows.

Future IP networks are required to support services with different and distinct end-to-end Quality of Service (QoS) requirements. Adaptive bandwidth provisioning serves as an attractive solution for providing guaranteed distinctive QoS and maintaining high network efficiency at the same time. However, most previous research on adaptive bandwidth provisioning is limited to the single-link case and assumes dedicated bandwidth. This paper studies multiple-link adaptive bandwidth provisioning for end-to-end statistical QoS guarantee in bandwidth sharing networks with Generalized Processor Sharing (GPS) schedulers. A feedback control model for end-to-end multiple-link adaptive bandwidth provisioning systems is presented. This model is verified by simulations, and it is shown to match the actual dynamics of adaptive bandwidth provisioning systems well. Based on this feedback control model, different controllers are designed and analyzed using control theory, and their performances are compared. The analysis and simulations show that the proposed end-to-end multiple-link bandwidth provisioning scheme is able to provide guaranteed end-to-end statistical QoS, and that both the adaptive P controller and adaptive PI controller can achieve better performance than the simple non-adaptive P controller.

As the Internet grows in scale and complexity, the benefits of network performance measurements and monitoring are significantly increasing. Sampling-based measurement methods provide adequate techniques for reducing the quantity of control data and attract growing interests. The research reported in this paper, addresses the issue of how to carry out the sampling in an adaptive fashion, so that the accuracy for measuring the quality of service parameters (delay, loss, jitter, throughput) is better if we know something about the traffic type and traffic parameters. Our study proposes and investigates a mechanism that can be set up to adaptively adjust the parameters of the sampling technique. Two realistic network topologies based on MPLS networks are setup to evaluate the proposed adaptive sampling scheme for monitoring and measuring network performance metrics. Compared with conventional sampling techniques (systematic and stratified sampling), simulation results are presented to illustrate that adaptive sampling provides the potential for better monitoring, control, and management of high-performance networks with higher accuracy.

Understanding of application traffic characteristics for distributed element management in SONET/SDH networks is key to provisioning of adequate resources so as to meet the application's quality of service requirements. This quality of service, defined as delay, loss, and throughput requirements, is presented based on measurements from customer networks. New traffic requirements resulting from evolution of transport products to higher payload capacities are also identified, along with performance road blocks which limit the evolution of distributed network management. Technologies and protocols which will alleviate these limitations are discussed along with a new resource allocation scheme to meet the diverse set of traffic conditions and quality of service requirements. Network interoperability in the context of global networks and from a performance perspective is then explored. Some form of common traffic descriptors seem to be necessary going beyond the current standardization of physical layer SONET/SDH overhead channels.

The introduction of broadband ATM networks and services will raise new operational issues. An area of significant challenge to the network operators will be network dimensioning. ATM networks differ from today's STM networks in that they will have to provide acceptable performance with respect to not only call blocking, but also cell loss and cell delay. Consequently, it is important to understand the interaction of call blocking and cell loss/delay in ATM networks. The paper presents a method of exploiting the interaction of call blocking and cell loss performance in order to efficiently operate the ATM networks. Specifically, the paper demonstrates that when the network is dimensioned or engineered to meet an appropriate call blocking objective, the cell loss performance perceived by the accepted calls can be significantly better than the cell loss objective set for the connection admission control (CAC) to admit or deny a call.>

Isopreference tests were used to study the subjective effects of channel transmission errors on computer-simulated PCM and previous-sample feedback DPCM voice systems. Numerically scaled isopreference contours were obtained and plotted on planes having as axes the number of quantization bitsNand the bit error probabilitypof the binary symmetric transmission channel. For anyNandpconsidered, the quality of DPCM speech was found to exceed that of PCM speech. WithNfixed andpdecreasing, the improvement of DPCM relative to PCM increased to the point where quantization noise limited the performance of both systems. Two nonredundant codes, natural binary and folded binary, were used to encode the quantizer output. The two codes yielded comparable speech quality. The maximum subjective quality obtainable for white Gaussian and Rayleigh fading channels was determined as a function of channel capacity. Use of optimum channel encoding was shown to reduce considerably the channel capacity required to obtain a given speech quality. The subjective ratings were compared with measured system signal-to-noise ratios. Finally, the effect of the overload level of the quantizer on the system signal-to-noise ratio was determined.