We demonstrated that load balancing using actual subscriber extension numbers was practical and effective against traffic congestion after a disaster based on actual data. We investigated the ratios of the same subscriber extension numbers in each prefecture and found that most of them were located almost evenly all over the country without being concentrated in a particular area. The ratio of every number except for the fourth-last digit in the last group of four numbers in a telephone number was used almost equally and located almost evenly all over the country. Tolerance against overload in the last, second-, and third-last single digits stays close to that in the ideal situation if we assume that each session initiation protocol server has a capacity in accordance with the ratio of each number on every single digit in the last group of four numbers in Japan. Although tolerance against overload in double-, triple-, and quadruple-digit numbers does not stay close to that in the ideal situation, it still remains sufficiently high in the case of double- and triple-digit numbers. Although tolerance against overload in the quadruple-digit numbers becomes low, disaster congestion is still not likely to occur in almost half of the area of Japan (23 out of 47 prefectures).

The widespread adoption of IP-based telecommunication core networks is leading to a paradigm shift in international interconnection where the traditional Time-Division Multiplexing (TDM) interconnection between telecommunication networks is being replaced by IP interconnection. IP eXchange (IPX) is an emerging paradigm in international IP interconnection that has novel requirements, such as an end-to-end Quality of Service (QoS) guarantee across multiple carriers. IPX is a future direction for international telecommunications, but it is not easy to understand the overall concept of IPX because it is derived from a wide variety of services, technical knowledge, and telecommunication backgrounds. The confusion and complexity of the technical elements hinder the development of IPX. Thus, this paper clarifies the state-of-the-art technical elements from an IPX perspective and discusses ongoing challenges and emerging services on IPX, particularly end-to-end QoS, Voice over IP issues, IP Multimedia Subsystem (IMS) interworking, and Long Term Evolution (LTE) roaming. This paper also surveys published academic research studies that were not focused primarily on IPX but which are likely to provide potential solutions to the challenges.

In IEEE 802.16e, power saving is one of the important issues for battery-powered mobile stations (MSs). We present a performance analysis of power saving class (PSC) of type I in IEEE 802.16e standard for voice over Internet protocol (VoIP) service with silence suppression in two-way communication. On-off pattern of a voice user in two-way communication is characterized by the modified Brady model, which includes short silence gaps less than 200 ms and talkspurt periods shorter than 15 ms, and so differs from the Brady model. Our analysis of PSC I follows the standard-based procedure for the deactivation of the sleep mode, where a uplink packet arrival during a mutual silence period wakes up the MS immediately while a downlink packet arrival waits to be served until the next listening window. We derive the delay distribution of the first downlink packet arriving during a mutual silence period, and find the dropping probability of downlink packets since a voice packet drops if it is not transmitted within maximum delay constraint. In addition, we calculate the average power consumption under the modified Brady model. Analysis and simulation results show that the sleep mode operation for the MS with VoIP service yields 3239% reduction in the power consumption of the MS. Finally we obtain the optimal initial/final-sleep windows that yield the minimum average power consumption while satisfying QoS constraints on the packet dropping probability and the maximum delay.

Providing quality-of service (QoS) guarantees in VoIP applications has become an urgent demand in wireless and mobile networks. One of the important issues is to find a simple quality metric fitted to low power mobile devices such as smart phones. This paper considers the gap ratio (the proportion of the accumulated gap periods over the whole call session) as a simple quality metric. Our study aims to find the optimum packet count threshold between two adjacent lost packets (referred to as Gmin in RTCP-XR), which is needed for the purpose of identifying whether the current packet at the receiver belongs to the gap state or the burst state, because quality prediction errors depend on the Gmin value when the gap ratio is used as a simple quality metric. Based on this metric, we propose an accounting model that can be a candidate accounting metric useful for a quality-based accounting mechanism.

In this article, we study different methods to enhance the capacity of Voice-over-IP (VoIP) in Evolved UTRAN (E-UTRAN) downlink. According to previous E-UTRAN system level simulation results, VoIP capacity with dynamic scheduling and semi-persistent scheduling is limited by available Physical Downlink Control CHannel (PDCCH) resources and Physical Downlink Shared CHannel (PDSCH) resources respectively. The different behavior of these two packet scheduling schemes and specific bottleneck on capacity lead us to formulate two distinctly different performance enhancement strategies, which are the main focus of this paper. For dynamic scheduling, we propose a bi-directional power control algorithm to rationalize PDCCH resource allocation, thus greatly improving VoIP capacity. For semi-persistent scheduling, we introduce Adaptive Transmission Bandwidth (ATB) to rationalize PDSCH resource allocation and further apply dynamic packet bundling in highly-loaded network, thus improving VoIP capacity in a considerable manner. The proposed enhancement methods are validated through large-scale system level simulations and the obtained system simulation results further confirm effectiveness of these enhancement methods.

The VoIP-based Internet Phonesystem is now seen as one of the killer applications in the high speed and broadband internet environment. Given the wide-spread use of the Internet Phone, it is necessary to provide security services for guaranteeing users' privacy. However, providing security service in Internet Phone has the possibility of incurring additional overheads such as call setup delay time. In this paper, we present a one-way key agreement model based on VoIP in order to reduce call setup time as well as protecting user privacy. The proposed approach decreases the delay time of the call setup in comparison with the previous models because our model enables the key generation in caller side without waiting the response from the receiver.

We present the development of a VoIP quality of service (QoS) measurement system that enables operators to diagnose a QoS degradation segment. Our system uses a flow-based passive measurement method to fulfill the requirement for QoS measurement in large-scale IP networks. In particular, we adopt an access control list (ACL)-based filtering function that selects traffic to monitor and develop a function for correlating signals and media data records. This correlation function is required to dynamically configure ACL-based filtering for monitoring media streams whose port numbers are determined by a signaling protocol. To improve the scalability of existing measurement systems, we also develop a hardware-based filtering engine on a commercial switch as well as a mediation box that performs QoS calculation based on traffic records exported by the engine in a distributed manner. We demonstrate the feasibility of the measurement system by evaluating a prototype system.

An optimal selection criterion of the modulation and coding scheme (MCS) for maximizing spectral efficiency is proposed in consideration of the signaling overhead of mobile WiMAX systems with a hybrid automatic repeat request mechanism. A base station informs users about the resource assignments in each frame, and the allocation process generates a substantial signaling overhead, which influences the system throughput. However, the signaling overhead was ignored in previous MCS selection criteria. In this letter, the spectral efficiency is estimated on the basis of the signaling overhead and the number of transmissions. The performance of the proposed MCS selection criterion is evaluated in terms of the spectral efficiency in the mobile WiMAX system, with and without persistent allocation.

Broadcasting information to users about new resource assignments generates a substantial mapping overhead. The mapping overhead influences the system throughput and, in particular, seriously affects the performance of voice-over-Internet protocol (VoIP) services. Persistent scheduling was introduced to reduce the mapping overhead. However, up to now no studies have mathematically analyzed the performance of the persistent scheduling. This paper develops analytical and simulation models and evaluates the performance of the persistent scheduling for VoIP services in mobile WiMAX systems.

This paper proposes a cross layer wireless VoIP service which integrates an Adaptive QoS Playout (AQP) algorithm, E-model, Stream Control Transmission Protocol (SCTP), IEEE 802.21 Media Independent Handover (MIH) middleware and two user motion detection services. The proposed AQP algorithm integrates the effect of playout control and lost packet retransmission based on the E-model. Besides, by using the partial reliable transmission service from SCTP and the handoff notification from MIH services in a cross layer manner, AQP can reduce the lateness loss rate and improve speech quality under high frame error rates. In the simulations, the performance of AQP is compared with a fixed playout algorithm and four adaptive playout strategies. The simulation results show that the lateness loss rate of AQP is 2% lower than that of existing playout algorithms and the R-factor is 16% higher than the compared algorithms when a network has 50 ms wired propagation delay and 2.5% frame error rate.

We present a session initiation protocol (SIP) network design for a voice-over-IP network to prevent congestion caused by people calling friends and family after a disaster. The design increases the capacity of SIP servers in a network by using all of the SIP servers equally. It takes advantage of the fact that equipment for voice data packets is different from equipment for signaling packets in SIP networks. Furthermore, the design achieves simple routing on the basis of telephone numbers. We evaluated the performance of our design in preventing congestion through simulation. We showed that the proposed design has roughly 20 times more capacity, which is 57 times the normal load, than the conventional design if a disaster were to occur in Niigata Prefecture struck by the Chuetsu earthquake in 2004.

In recent years elephant flows are increasing by expansion of peer-to-peer (P2P) applications on the Internet. As a result, bandwidth is occupied by specific users triggering unfair resource allocation. The main packet-scheduling mechanism currently employed is first-in first-out (FIFO) where the available bandwidth of short flows is limited by elephant flows. Least attained service (LAS), which decides transfer priority of packets by the total amount of transferred data in all flows, was proposed to solve this problem. However, routers with LAS limit flows with large amount of transferred data even if they are low-rate. Therefore, it is necessary to improve the quality of low-rate flows with long holding times such as voice over Internet protocol (VoIP) applications. This paper proposes rate-based priority control (RBPC), which calculates the flow rate and control the priority by using it. Our proposed method can transfer short flows and low-rate flows in advance. Moreover, its fair performance is shown through simulations.

VoIP service is expected as one of the key applications of Mobile WiMAX, but the speech quality of VoIP service often suffers deterioration due to the fluctuating transmission delay called jitter. This is commonly ameliorated by a de-jitter buffer, and we aim to find the optimal size of de-jitter buffer to achieve speech quality comparable to PSTN. We developed a new model of the packet drops at the de-jitter buffer and the end-to-end packet delay which takes account of the additional delay introduced by the WiMAX power-saving mode. Using our model, we analyzed the optimal size of the de-jitter buffer for various network parameters, and showed that the results obtained by analysis accord with simulation results.

With the wide expansion of voice services over the IP networks (VoIP), the volume of this delay sensitive traffic is steadily growing. The current packet schedulers for IP networks meet the delay constraint of VoIP traffic by simply assigning its packets the highest priority. This technique is acceptable as long as the amount of VoIP traffic is relatively very small compared to other non-voice traffic. With the notable expansion of VoIP applications, however, the current packet schedulers will significantly sacrifice the fairness deserved by the non-voice traffic. In this paper, we extend the conventional Deficit Round-Robin (DRR) scheduler by including a packet classifier, a Token Bucket and a resource reservation scheme and propose an integrated packet scheduler architecture for the growing VoIP traffic. We demonstrate through both theoretical analysis and extensive simulation that the new architecture makes it possible for us to significantly improve the fairness to non-voice traffic while still meeting the tight delay requirement of VoIP applications.

Switching a communication path from one Access Point (AP) to another in inter-domain WLANs is a critical challenge for delay-sensitive applications such as Voice over IP (VoIP) because communication quality during handover (HO) is more likely to be deteriorated. To maintain VoIP quality during HO, we need to solve many problems. In particular, in bi-directional communication such as VoIP, an AP becomes a bottleneck with the increase of VoIP calls. As a result, packets queued in the AP buffer may experience a large queuing delay or packet losses due to increase in queue length or buffer overflow, thereby causing the degradation of VoIP quality for the Mobile Nodes (MNs) side. To avoid this degradation, MNs need to appropriately and autonomously execute HO in response to the change in wireless network condition, i.e., the deterioration of wireless link quality and the congestion state at the AP. In this paper, we propose an HO decision strategy considering frame retries, AP queue length, and transmission rate at an MN for maintaining VoIP quality during HO. Through simulation experiments, we then show that our proposed method can maintain VoIP quality during HO by properly detecting the wireless network condition.

This paper proposes bidirectional packet aggregation and coding (BiPAC), a packet mixing technique which jointly applies packet aggregation and network coding in order to increase the number of supportable VoIP sessions in wireless multi-hop mesh networks. BiPAC applies network coding for aggregated VoIP packets by exploiting bidirectional nature of VoIP sessions, and largely reduces the required protocol overhead for transmitting short VoIP packets. We design BiPAC and related protocols so that the operations of aggregation and coding are well-integrated while satisfying the required quality of service by VoIP transmission, such as delay and packet loss rate. Our computer simulation results show that BiPAC can increase the number of supportable VoIP sessions maximum by around 87% as compared with the case when the packet aggregation alone is used, and 600% in comparison to the transmission without aggregation/coding. We also implement BiPAC in a wireless testbed, and run experiments in an actual indoor environment. Our experimental results show that BiPAC is a practical and efficient forwarding method, which can be implemented into the current mesh hardware and network stack.

Overlay networks which are dynamically created over underlying IP networks are becoming widely used for delivering multimedia contents since they can provide several additional user-definable services. Multiple overlay paths between a source-destination overlay node pair are designed to improve service robustness against failures and bandwidth fluctuation of the underlying networks. Multimedia traffic can be distributed over those multiple paths in order to maximize paths' utilization and to increase application throughputs. Most of flow-based routing algorithms consider only common metrics such as paths' bandwidth or delay, which may be effective for data applications but not for real-time applications such as Voice over IP (VoIP), in which different levels of such performance metrics may give the same level of the performance experienced by end users. This paper focuses on such VoIP overlay networks and proposes a novel alternative path based flow routing algorithm using an application-specific traffic metric, i.e. "VoIP Path Capacity (VPCap)," to calculate the maximum number of QoS satisfied VoIP flows which may be distributed over each available overlay path at a moment. The simulation results proved that more QoS-satisfied VoIP sessions can be established over the same multiple overlay paths, comparing to traditional approaches.

In this paper, we analyze the extended real-time Polling Service (ertPS) algorithm in IEEE 802.16e systems, which is designed to support Voice-over-Internet-Protocol (VoIP) services with data packets of various sizes and silence suppression. The analysis uses a two-dimensional Markov Chain, where the grant size and the voice packet state are considered, and an approximation formula for the total throughput in the ertPS algorithm is derived. Next, to improve the performance of the ertPS algorithm, we propose an enhanced uplink resource allocation algorithm, called the e 2rtPS algorithm, for VoIP services in IEEE 802.16e systems. The e 2rtPS algorithm considers the queue status information and tries to alleviate the queue congestion as soon as possible by using remaining network resources. Numerical results are provided to show the accuracy of the approximation analysis for the ertPS algorithm and to verify the effectiveness of the e 2rtPS algorithm.

This paper proposes a VoIP (Voice over Internet Protocol) session capacity expansion method that uses periodic packet transmission suppression control for wireless LANs. The proposed method expands the VoIP session capacity of an AP without critically degrading the QoS (Quality of Service) of all stations. Simulation results show the proposed method with 0.5% packet suppression control on each station expands a VoIP session capacity by up to 5% compared to a legacy method while satisfying required QoS for all stations.

We propose an architecture of Intrusion Detection System (IDS) for VoIP using a protocol specification-based detection method to monitor the network traffics and alert administrator for further analysis of and response to suspicious activities. The protocol behaviors and their interactions are described by state machines. Traffic that behaves differently from the standard specifications are considered to be suspicious. The IDS has been implemented and simulated using OPNET Modeler, and verified to detect attacks. It was found that our system can detect typical attacks within a reasonable amount of delay time.