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Network programmability is a key feature of fifth generation (5G) system which, in combination with cloud-based services, can support many use cases, including mission critical and healthcare communications. Programmability enables flexibility in customization of service connectivity. Multi-access Edge Computing (MEC) services and applications are enablers for network programmability. In this paper, MEC capabilities for programmability of multiparty multimedia call control at the network edge are studied. Multiparty video calls are one of the key applications of 5G, and are efficient way to exchange ideas, knowledge, expertise, information, and so on. The paper presents an approach to design MEC Application Programming Interfaces (APIs) which enable third party applications to create multiparty multimedia sessions and dynamically manage session participations. The API functionality is described by required information and message flows. The paper specifies the proposed MEC API with data model. Feasibility study includes modelling and formal validation of multiparty session state models supported by the network and mobile edge application. The latency injected by the API is evaluated by emulation.

The COVID-19 pandemic necessitated stringent visitor restrictions in critical care departments worldwide, creating challenges in keeping family members connected to patients and clinical staff. Previous studies have examined how hospitals addressed this challenge by repurposing existing tele-ICU systems or by using personal smartphones as a workaround and have analyzed clinical and family feedback. This case report addresses the experience of rapidly implementing a video-call system in the critical care department of a tertiary referral hospital that had no prior video-call system in place, detailing the key requirements in that setting. The 24 requirements were identified via interviews and surveys to both clinical and technical professionals. The top requirements identified were sound and video quality, usability for clinical staff, call control by staff, and patient privacy. From tailoring a video-call solution for this setting, we learned that video-endpoint selection is a key design decision. The initial proposal was to use wireless tablets, but the selection of a large wired video-endpoint allowed us to better address the requirements in the critical care setting. This was based on several characteristics of the large wired video-endpoint, including: high-fidelity video and sound, with directional noise-cancelling; large touch-screen setup for minimal-click navigation; wired as well as wireless connectivity.

International audience; We consider a blended call center with calls arriving over time and an infinitely backlogged amount of outbound jobs. Inbound calls have a non-preemptive priority over outbound jobs. The inbound call service is characterized by three successive stages where the second one is a break; i.e., there is no required interaction between the customer and the agent for a non-negligible duration. This leads to a new opportunity to efficiently split the agent time between inbound calls and outbound jobs. We focus on the optimization of the outbound job routing to agents. The objective is to maximize the expected throughput of outbound jobs subject to a constraint on the inbound call waiting time. We develop a general framework with two parameters for the outbound job routing to agents. One parameter controls the routing between calls and the other performs the control inside a call. We then derive structural results with regard to the optimization problem and numerically illustrate them. Various guidelines to call center managers are provided. In particular, we prove for the optimal routing that at least one of the two outbound job routing parameters has an extreme value.

With the telecom market reaching saturation in many regions and revenues from voice calls decreasing, telecom operators are attempting to identify new sources of revenue. For this purpose, these operators can take advantage of their core functionalities, such as location and call control, by exposing them as services to be composed by developers with third-party offerings available over the web. Mashups enable technology to compose those services into new applications. However, existing mashup tools have some limitations in such convergent web-service-based integration, such as extra programming efforts for ordinary users. To address these limitations, we propose a novel LSMP, which includes the SDM, SRM, SCM, and SEM. The SDM defines a unified structure for atom-services to reduce the development complexity. The SRM defines a structurized data flow to reduce the coupling of data transfer between atom-services. The SCM provides a design method for the service process, with key data as the driving characteristic. The SEM provides a container for the service logic execution. With these models, ordinary users can create new personalized services freely via an intuitive and easy-to-use UI on a web browser.

The rapid growth in subscribers and usage of multimedia services enlarges the volume of Session Initiation Protocol (SIP) call control signalling creating a need to understand Quality of Experience (QoE) in this case and improve it. This paper provides an analysis of influence of SIP call control signalling on QoE for Voice over Internet Protocol (VoIP) service. The aim was to investigate whether SIP call control signalling load has the influence on the human perception of SIP signalling performances and QoE, and to identify the importance of distinct SIP-based signalling performance metrics. Moreover, the intention was to determine whether SIP call control signalling load changes its impact if previously proposed algorithm for differentiated treatment of SIP messages is activated, and quantify mutual relationships of considered user perceptions and QoE. The findings show that SIP call control signalling load has a strong and negative impact on dependent variables and that the proposed algorithm improves QoE and human perception of SIP signalling performances.

Traditional PSTN-based Call-Center fail to meet agent distribution deployments and lower operating costs. With the development of NGN technology, Soft-Switch-based Call-Center has solved the pain points of enterprises. This paper analyzes the special needs of distributed Call-Center system, and on the basis of analyzing and studying Soft-Switch technology, the platform architecture of distributed Call-Center system based on VoIP is given. This paper focuses on the support process of SIP protocol for call control function of Call-Center system, as well as the key equipment in Soft-Switch system. Finally, a practical Soft-Switch system implementation scheme is introduced in conjunction with the project.

Significant research efforts for the convergence of web and telecommunication services have been recently spent by research and industry stakeholders. The IETF and W3C are cooperating in specifying how web browsers should evolve to natively support communication services. In this perspective, devising novel mechanisms for signaling message exchange and possible interworking between Web- and SIP-based systems is a hot topic of research. Indeed, discussions are still ongoing on how differences between REpresentational state transfer (REST) and session initiation protocol (SIP) models should be coped with. This issue is made more difficult by the lack of rigorous modeling of RESTful systems. In this paper we propose a rigorous approach for design and implementation of REST communication services (e.g., a call service) which leverages formal verification techniques, while allowing to meet a specific performance requirement (i.e., maximum call setup delay). First, we formalize the call resource behavior through a Finite State Machine representation by modeling and simulating service expected behavior and its interworking with SIP User Agents through a tool for the analysis of communicating state machines. Then, we use the model-checking capabilities offered by the tool for the verification of formal properties. Finally, we implement a prototype that, thanks to the previous formalization step, is shown to be functionally correct, while yielding acceptable performance.

Purpose– The purpose of this paper is to investigate how customer involvement in call routing affects the internal operations of the call centre service system by examining customer usability problems with the interactive voice response (IVR) system and the practices of agents used to redirect incorrectly routed calls.Design/methodology/approach– A qualitative case study combined direct observation of live calls through sit-bys with agents and semi-structured interviews conducted with coaches and managers within 13 separate teams across all four functional areas of a call centre operation.Findings– Customer use of the IVR system involved effort, capability, and arrival forms of customer-induced variability, which produced incorrect call inputs into the call centre. Shared norms and attitudes concerning knowledge, IT use, and responsibility for different call types within teams were associated with redirecting practices which lead to the problematic rerouting of calls. Problems with call routing and rerouting negatively affected operational efficiency and undermined customer satisfaction.Research limitations/implications– The paper is based upon a single case study so further research is required to examine how problems identified are manifest in call centre operations of different size and complexity. The qualitative approach develops rich insights but these findings would benefit from a quantitative focus in the future.Practical implications– The customer experience of IVR systems should be continually monitored to identify usability problems and ensure effective design, while call centre management should attempt to increase teams’ awareness of and ability to successfully redirect incorrectly routed calls.Originality/value– The paper conceptualises the mutual influence of macro-level service system design and the micro-level behaviour of customers and agents upon each other. In practice, formal design decisions such as input uncertainty, decoupling, and interdependence patterns are continually reproduced or modified. Shared attitudes and norms of teams and their behavioural influence upon agents’ call handling practices are identified as a cause of coordination problems in call centres service systems. Internal rerouting by agents is also identified as a crucial operational process and important area for future research.

This study examines how telephone conversations are transferred and re-opened in calls to the Swedish Board for Student Support. The study relates to research on the opening and closing of telephone conversations, and how with each transfer, the reason for the call may be new to one participant, but may not be new for the other participant. The data consist of 126 calls in Swedish where transfer is explicitly oriented towards by at least one of the participants. The analysis shows that call-takers take great care to ensure that callers know that they are being transferred. Some call-takers also take care to frame the re-opened call as transferred, others do not; in the case of the latter, the interactional burden of establishing a common frame for the call as transferred falls on the caller.

Algorithmica, 47 (3), ISSN:0178-4617, ISSN:1432-0541

The emergency call device has been installed and operated for the elevator passenger safety, but the call can not be connected or delay to the operator in an emergency. The application of the Internet of Things(IoT) technology is required for the systematic management and reliability of the emergency call device installed in the elevator. To meet these requirements it is necessary to apply the RMS(Remote Management System). Emergency call device periodically reports the operating status information and the emergency call history information of the emergency call device to the RMS in the PSTN(Public Switched Telephone Network) environment, and transmits the call request information in an emergency. RAS(Remote Access Server) plays a role of converting the PSTN signals to TCP / IP. RMS reports a significant outbreak call and abnormal operating conditions to administrators. In this study, we developed optimized protocol to PSTN environment that is required by RMS and implemented its differentiated process. And a RAS-RMS Protocol was added to the Application Layer. Development protocol consisting of simple structure as compared to CoAP, MQTT or HTTP can work properly in low-spec CPU and transmit elevator emergency key information easily.

The evolution towards mobile networks flat architecture entreat a key-role for IP mobility management in providing the ubiquitous always-on network access services. This paper provides prospects for efficient mobility management in SDN plus mobile network architecture and describe important call control flows in inter system handover scenario. Distribution of gateway function approach has been followed and evolved with SDN technology. Key improvements with proposed architecture are to support seamless mobility in heterogeneous access environments, remove the chains of IP preservation and optimal data path management according to application needs. The paper assays the proposed evolution in terms of numbers of signaling messages processed by control entities for an inter system handover procedure relative to current mobile network architecture.

SummaryBackground: Discovery of clinical workflows to target for redesign using methods such as Lean and Six Sigma is difficult. VoIP telephone call pattern analysis may complement direct observation and EMR-based tools in understanding clinical workflows at the enterprise level by allowing visualization of institutional telecommunications activity.Objective: To build an analytic framework mapping repetitive and high-volume telephone call patterns in a large medical center to their associated clinical units using an enterprise unified communications server log file and to support visualization of specific call patterns using graphical networks.Methods: Consecutive call detail records from the medical center’s unified communications server were parsed to cross-correlate telephone call patterns and map associated phone numbers to a cost center dictionary. Hashed data structures were built to allow construction of edge and node files representing high volume call patterns for display with an open source graph network tool.Results: Summary statistics for an analysis of exactly one week’s call detail records at a large academic medical center showed that 912,386 calls were placed with a total duration of 23,186 hours. Approximately half of all calling called number pairs had an average call duration under 60 seconds and of these the average call duration was 27 seconds.Conclusions: Cross-correlation of phone calls identified by clinical cost center can be used to generate graphical displays of clinical enterprise communications. Many calls are short. The compact data transfers within short calls may serve as automation or re-design targets. The large absolute amount of time medical center employees were engaged in VoIP telecommunications suggests that analysis of telephone call patterns may offer additional insights into core clinical workflows.Citation: Rucker DW. Using telephony data to facilitate discovery of clinical workflows. Appl Clin Inform 2017; 8: 381–395 https://doi.org/10.4338/ACI-2016-11-RA-0191

This article shows the usage and the experimental analysis of open source software packages for designing and implementing carrier-grade multimedia services. The shown implementation is based on the use of Java Call Control APIs for managing SIP sessions and MGCP connections. This JCC implementation has been integrated within an open source platform, Mobicents JSLEE. Mobicents is a runtime environment suitable for deploying services integrating voice, video, and messaging over different networks. An extensive experimental campaign has allowed suitable deployment strategies to be identified, depending on the service features. The main contribution of this article is to present these strategies, consisting of resource allocation and configuration in a virtualized environment. The results of our experiments are aimed at providing general guidance on configuring servers in a virtualized environment such as a cloud. They can also be helpful to address the server placement problem, which is beyond the scope of this article. An additional contribution of this article consists of a simple and effective model to predict the achievable server throughput, expressed in calls per second. For this purpose, we have measured the CPU time for processing SIP and MGCP messages, and estimated the maximum sustainable load of each virtual machine. In this way, we can preliminarily estimate the number of virtualized instances of the platform needed for handling a given workload with the desired quality of service.

Call admission control is a very important measure in CDMA system to guarantee the quality of the communicating links. In future wireless networks multimedia traffic will have different QoS requirements. The call admission control is very important measure in CDMA system to guarantee the quality of the communicating link In future wireless networks multimedia traffic will have different QoS requirements. We have proposed call admission and call control algorithms for QoS provisioning for multimedia traffic in a cellular mobile network. In this paper we propose a call admission control algorithm that keeps the dropping probability below a specified level while achieving high channel utilization.

Nowadays a lot of different researches are performed based on call duration distributions (CDD) analysis. However, the majority of studies are linked with social relationships between the people. Therefore the scarcity of information, how the call duration is associated with a user's location, is appreciable. The goal of this paper is to reveal the ties between user's voice call duration and the location of call. For this reason we analyzed more than 5 million calls from real mobile network, which were made over the base stations located in rural areas, roads, small towns, business and entertainment centers, residential districts. According to these site types CDD’s and characteristic features for call durations are given and discussed. Submitted analysis presents the users habits and behavior as a group (not an individual). The research showed that CDD’s of customers being them in different locations are not equal. It has been found that users at entertainment, business centers are tend to talk much shortly, than people being at home. Even more CDD can be distorted strongly, when machinery calls are evaluated. Hence to apply a common CDD for a whole network it is not recommended. The study also deals with specific parameters of call duration for distinguished user groups, the influence of network technology for call duration is considered.

In a call center, agents may handle calls at different speeds, and also may be more or less successful at resolving customers' inquiries, even when only considering customers calling with similar requests. One common measure of successful call resolution is whether or not the call results in the customer calling back. This presents a natural trade-off between speed and quality, where speed is defined as the average time before an incoming call is answered (the average waiting time) and quality is defined as the percentage of all arriving calls that do not result in callbacks (the call resolution). The relevant control is the routing, that is, the decision concerning which agent should handle an arriving call when more than one agent is available. In an inverted-V model setting, we formulate an optimization problem with the dual performance objective of minimizing average customer waiting time and maximizing the call resolution. We solve this optimization problem asymptotically in the Halfin–Whitt many-server limit regime, interpret its solution as a routing control for the discrete-event system, and show via simulation that the interpreted routing control is on the efficient frontier. In particular, any routing control that has a lower average waiting time (higher call resolution) must also have a lower call resolution (higher average waiting time).

To improve the network of Beijing 12316 voice call system and access the flexibility, enhance scalability and maintainability of the system, this paper analyzes technical advantages of the IP call system compared with the traditional voice call system, design and implement the voice call system, which is based on H.323 protocol IP on original Beijing 12316 voice call system. The system contains interface layer, traffic flow control layer and encapsulation layer, and the system is based on communication and control functions of H.323 protocol, which is invoked in the form of ActiveX controls, which can be called directly to complete the basic business functions. Users only need a computer with audio equipment or IP telephone of network to use the system, which will improve service quality of the Beijing 12316 IP call system greatly.

In this paper we consider the influence of a call forwarding service on the traffic process in modern telecommunication networks. We analyse in detail only the case when the called user is busy. It is proved that call forwarding not only increases utilization of servers and the percent of successful calls but it also increases the call loss. On the simple example, we showed that this call loss increase is greater in the case of local (internal) and incoming calls, but smaller in the case of outgoing calls. The reason for such behaviour is in the role of call forwarding function in the case of internal and incoming traffic. In that situation call forwarding function decreases the effect of limited number of users on the decrease of offered traffic, comparing to the case of internal and incoming traffic without call forwarding (Engset traffic model). This statement is illustrated by comparative graphics of traffic loss without call forwarding function, and with this function when considering, separately, internal, incoming, and outgoing traffic. [Projekat Ministarstva nauke Republike Srbije, br. TR32007: Multiservice optical transport platform with OTN/40/100 Gbps DWDM/ROADM and Carrier Ethernet functionality]

Generally in call centers, voice calls (say Type 1 calls) are given higher priority over e-mails (say Type 2 calls). An arriving Type 1 call has a preemptive priority over a Type 2 call in service, if any, and the preempted Type 2 call enters into a retrial buffer (of finite capacity). Any arriving call not able to get into service immediately will enter into the pool of repeated calls provided the buffer is not full; otherwise, the call is considered lost. The calls in the retrial pool are treated alike (like Type 1) and compete for service after a random amount of time, and can preempt a Type 2 call in service. We assume that the two types of calls arrive according to a Markovian arrival process (MAP) and the services are offered with preemptive priority rule. Under the assumption that the service times are exponentially distributed with possibly different rates, we analyze the model using matrix-analytic methods. Illustrative numerical examples to bring out the qualitative aspects of the model under study are presented.

In many call centers, agents are trained to handle all arriving calls but exhibit very different performance for the same call type, where we define performance by both the average call handling time and the call resolution probability. In this paper, we explore strategies for determining which calls should be handled by which agents, where these assignments are dynamically determined based on the specific attributes of the agents and/or the current state of the system. We test several routing strategies using data obtained from a medium-sized financial service firm's customer service call centers and present empirical performance results. These results allow us to characterize overall performance in terms of customer waiting time and overall resolution rate, identifying an efficient frontier of routing rules for this contact center.

Wireless communication networks based on frequency division multiplexing (FDM in short) play an important role in the field of communications, in which each request can be satisfied by assigning a frequency. To avoid interference, each assigned frequency must be different from the neighboring assigned frequencies. Since frequencies are scarce resources, the main problem in wireless networks is how to fully utilize the given bandwidth of frequencies. In this paper, we consider the online call control problem. Given a fixed bandwidth of frequencies and a sequence of communication requests arriving over time, each request must be either satisfied immediately after its arrival by assigning an available frequency, or rejected. The objective of the call control problem is to maximize the number of accepted requests. We study the asymptotic performance of this problem, i.e., the number of requests in the sequence and the bandwidth of frequencies are very large. In this paper, we give a 7/3-competitive algorithm, say CACO, for the call control problem in cellular networks, improving the previous 2.5-competitive result, and show that CACO is best possible among a class of HYBRID algorithms.

Binary CDMA is a new standard technology for wireless communication developed by our country that makes high speed communications and good quality of services. In this paper we researched the development methods and implementation of a wireless VoIP System based on the Binary-CDMA technology that makes it freely installed in any place without phone cables and laying works. We designed the structure of the Binary-CDMA Wireless VoIP System and defined messages and data structures for call control. And then we designed and implemented call control message flow between VoIP terminals for user`s communication.

Transmitting information via communicative signals is integral to interacting with conspecifics, and some species achieve this task by varying vocalizations to reflect context. Although signal variation is critical to social interactions, the underlying neural control has not been studied. In response to a predator, black-capped chickadees (Poecile atricapilla) produce mobbing calls (chick-a-dee calls) with various parameters, some of which convey information about the threat stimulus. We predicted that vocal parameters indicative of threat would be associated with distinct patterns of neuronal activity within brain areas involved in social behavior and those involved in the sensorimotor control of vocal production. To test this prediction, we measured the syntax and structural aspects of chick-a-dee call production in response to a hawk model and assessed the protein product of the immediate early gene FOS in brain regions implicated in context-specific vocal and social behavior. These regions include the medial preoptic area (POM) and lateral septum (LS), as well as regions involved in vocal motor control, including the dorsomedial nucleus of the intercollicular complex and the HVC. We found correlations linking call rate (previously demonstrated to reflect threat) to labeling in the POM and LS. Labeling in the HVC correlated with the number of D notes per call, which may also signal threat level. Labeling in the call control region dorsomedial nucleus was associated with the structure of D notes and the overall number of notes, but not call rate or type of notes produced. These results suggest that the POM and LS may influence attributes of vocalizations produced in response to predators and that the brain region implicated in song control, the HVC, also influences call production. Because variation in chick-a-dee call rate indicates predator threat, we speculate that these areas could integrate with motor control regions to imbue mobbing signals with additional information about threat level.

Call control allows an organism to produce an acoustic signal irrespective of its own underlying emotional state. It is thus a prerequisite to “higher” abilities, such as call imitation, innovation and the use of arbitrary or deceptive calls, and therefore to speech. However, among primates, call control is presumed to be greatly confined to humans (Seyfarth and Cheney 2008). Consequently, there is little agreement about its evolutionary precursors (Christiansen and Kirby 2003). Essentially two major models and lines of evidence have been proposed; speech evolved (1) as an extension of acoustic communication in non-human primates (e.g. Seyfarth et al.1980; Slocombe and Zuberbuhler 2005; Arnold and Zuberbuhler 2006; Wich et al.2009) or (2) from non-human primate gestural communication (e.g. Rizzolatti and Arbib 1998; Corballis 2003; Arbib Michael et al. 2008). These models have been seen as mutually exclusive or as sequential accounts in which calls replace gestures (Brown et al. 1999), however, both face limitations concerning the emergence of call control in our evolutionary lineage. Did call control derive from an essentially emotional call use, or from an essentially voluntary gesture use, as that of non-human primates? The acoustic model needs to explain how a fundamentally close-ended acoustic system became open-ended (i.e. with limitless number of elements; alike speech). The gestural model needs to clarify the behaviors and respective functional advantages that allowed a shift (or “translation”) from an open-ended gestural system to an open-ended acoustic system.

The study, one of the important solutions in the Call-center about the Call Management System Server and the Call Management System Server than that of the current situation and the advantages and disadvantages of using the measures studied and improved integration of future models of the proposed statistical solution.

two major standards in the multimedia services over IP area are the protocol suites H.323 (ITU-T) and SIP (IETF).Both have emerged as competing protocol standards for the signaling and call control of IP telephony. SIP is designed with a broader scope, offering functions specifically designed to enable easy extensions; it should be the advantage for new potential services. H.323 is still the more mature standard; H.323 provides better interoperability and interworking (PSTN, ISDN). We can assume a coexistence of both protocols. We compare voice performance parameters like talking time, total sessions, delay, jitter packet delivery ratio, call set up times and throughput, establish the differences in performance of these two VoIP protocols through an Qualnet simulation. SIP and H.323 are used for establishment and release of Voice over IP calls as well as Video and M edia calls. Both these protocols play a very important role in terms of optimizing the call set up time and call reliability and flexibility over IP networks for real time applications like voice and video. The obtained results are discussed to highlight the impact of both H.323 and SIP in ad- hoc networks. Keywordsover IP (VOIP), H.323, SIP, Call Set up Time, End to End delay, internet telephony, signaling Protocol.

In the last few years we have experienced a dramatic increase in the use of IP networks for voice applications (VoIP) over wireless networks due to increased bandwidth availability and enhanced device capabilities. Since demand often exceeds available capacity, Call Admission Control mechanisms are in place to prevent the uncontrolled usage of bandwidth. Through the use of an intermediary gateway, VoIP calls are in many cases terminated to a normal landline or cellphone; the capacity of such a gateway is also a finite resource since the number of users can vary significantly as many are mobile. In this article we propose an enhanced scheme that aims to manage access to the lines available so that they are used in a fair manner and utilized to the highest degree possible. This management is facilitated by enhancing a proxy implementation with a number of call scheduling policies. The ability to satisfy pending call requests as soon as lines become available, results in increased user service satisfaction. Moreover, it increases line utilization which is crucial from an economic viewpoint. The ultimate goal is to improve Quality of Experience which is deemed as highly important especially considering that wireless network users experience opportunistic and intermittent connectivity.

For Push-To-Talk (PTT) system based on Public Mobile Data Network (PMDN), the end-to-end time delay is the key aspect of the user’s experience. The Push-Over-Cellular (POC) scheme defined by Open Mobile Alliance (OMA) is based on the VoIP phone model and use SIP protocol as the call control scheme. The call setup time delay in SIP may reach to several seconds, which is unacceptable for the PTT service. In this paper, we provide a new call control scheme for PTT system based on PLMN network. By combining the apriority knowledge of PTT call model and the priority control scheme, we encapsulate the signaling message and the voice data into a same data packet, when the user push the button, the voice and the call control signaling are sent to the server at the same time. So the long time delay of call setup procedure of POC scheme can be eliminate. The end-to-end call delay can be decreased significantly. The experiment result based on the commercial CDMA2000 1X network of China Unicom shows that the call delay can be decreased to 600 ms, which approach to the traditional trunk communication system’s requirement.

This paper considers 'two-stage' call centers where some incoming calls are completed by first service while others require an additional second service. Although this type of call center is not uncommon, it has not been dealt with, if any, in the call center literature. In this paper, we introduce the concept of the 'two-stage' call center and discuss its features. Furthermore, we develop an effective outsourcing strategy in 'two-stage' call centers. To this end, we model 'two-stage' service system and propose several call routing structures. The structures are compared through numerical test and conventional queueing theories form the theoretical basis of our study.

The IP multimedia subsystem is becoming the de facto standard for IP-based multimedia services, while wireless sensor networks are gaining in popularity due to their ability to capture a rich set of contextual information. Integrating the sensing capabilities of WSNs in the IMS can open the door to a wide range of context-aware applications in areas such as wireless healthcare and pervasive gaming. We have previously proposed a presence-based architecture for WSN/IMS integration. This architecture relies on two key components: a WSN/IMS gateway acting as an interworking unit between WSNs and the IMS; and an extended presence server serving as a context information management node in the core network. In this article we focus on the design and implementation of these two components. Furthermore, two applications (a pervasive game and a personalized call control application) are used to concretely show how new applications can be developed using our architecture. Performance has also been evaluated. Several important findings were made in the course of this work; one is that the IMS integration with a large and evolving variety of WSNs may be a never-ending endeavor - the gateway requiring constant upgrading due to the lack of standard APIs for the interaction with sensors produced by different vendors. Another finding is that while the introduction of context as an application building block in the IMS ensures the availability of additional contextual information in the network and enables fast and easy development of context-aware applications, the lack of mature IMS application development toolkits remains a roadblock.

The Power Utility of Serbia Company intends to migrate portions of its time-division-multiplexing- based telephony network to IP. The high-level, capital project requirements include: (a) building a new VoIP infrastructure; (b) exchanging approximately one-third of the existing TDM-based PBXs with new VoIP based and hybrid PBXs; and (c) integrating these new PBXs with the remaining existing legacy PBXs to create a new hybrid network. Telephony service availability should be five nines (99.999 percent), a requirement that created a derived requirement for network element redundancy. One of the most important elements is the softswitch. The softswitch is the equipment responsible for call control of IP phones and voice gateways at the level of the Power Utility of Serbia Company telephone network. Softswitch redundancy in a failover configuration proved to be most challenging because of a performance requirement to achieve a switchover time of six seconds or less, whereas commercially available softswitches typically support switchover times of a few minutes. Another significant challenge was integrating the legacy PBXs and new (VoIP) PBXs. Although industry marketing promotes SIP trunking as the preferred integration/interconnection method, we found that with the exception of bearer traffic (voice media) pathing, service interoperability between the equipment of different vendors was often problematic. Vendors typically suggest Q.Sig as an interconnection alternative when SIP trunking is infeasible, but some vendor interoperability problems persisted. This article presents technical solutions to these major issues that can benefit similar TDM-to-IP migration and hybridization projects.

The evolved 3GPP system is a hybrid mobile network architecture supporting several radio access technologies and several mobility mechanisms. In this article we briefly review the architecture and key components of this system, with particular emphasis on how it can support voice call mobility in several deployment scenarios. First, we present the so-called single-radio voice call continuity mechanisms that enable mid-call handover of VoIP calls from E-UTRAN access to the legacy UTRAN/GERAN or lxRTT access. Then we focus on deployment scenarios that do not support voice services on E-UTRAN and present the so-called fallback mechanisms that enable handover from E-UTRAN to UTRAN/GERAN or lxRTT at the beginning of a voice call. Finally, we address the application- layer voice call handover mechanisms enabled by the IP multimedia subsystem. Our conclusion is that the next generation of 3GPP systems are highly sophisticated mobile communication systems that support extended voice call mobility mechanisms, capable of addressing all commercial deployment needs.

Recently, real-life data have revealed that the number of calls originating from or received by a telephone number in a network follows a power-law distribution. They show that a few telephone numbers make or receive a very large number of calls, whereas a large number of telephone numbers make or receive very few calls. The data have overthrown the general assumption that all telephone numbers are similar in generating telephone traffic. The first objective of this paper is to therefore construct a telephone call network (TCN) with connection properties following power-law distributions. With a more realistic TCN, researchers and engineers will be able to evaluate the telephone traffic behavior more accurately. Having constructed the aforementioned TCNs, we then consider the scenario when there is a sudden surge in the number of telephone calls, for example, during natural or man-made disasters. Under such a condition, the telephone network is usually overloaded and cannot operate properly. To mitigate the problem, we propose a preferential call blocking (PCB) scheme, aiming at blocking calls to target telephone numbers which have large numbers of incoming calls (in-strengths). We will investigate the effect on the carried traffic intensity when the PCB scheme is applied. We will compare the results with a benchmark, which corresponds to the case when all calls are blocked with equal probability. For the sake of completeness, we will also study the effectiveness of the blocking schemes when applied to a traditional TCN, in which all telephone numbers can call one another with equal probability.

Contact centers are complex call centers to handle large volume of inbound, outbound or both types of calls depending on the business purpose. Call centers assume the role of the primary contact medium for many companies from a wide range of industries with their customers or clients. Despite of being seen traditionally as adding cost to the companies' bottom lines, call centers are now viewed by many companies to turn a service request into an opportunity to sell additional products and services. This sales attempt is called cross-selling. The opportunity to generate profit from an existing customer-base is a key factor for a successful call center. This paper introduces a framework for balancing cross-selling and service activities in a call center setting from a queuing science point of view. The main goal of this study is to introduce a framework to maximize a call center's performance without degrading the service quality. Our framework is based on the usage of real-time queue characteristics, customer profile information and server-skill set information from a cross-sell point of view.

Existing mobile service providers allow their customers to receive telecommunications services when they roam to other countries. However, in the standard mobile call setup procedure, a call from the visited country to a roaming customer at that country results in two international calls. This phenomenon is referred to as the tromboning effect. In this paper, we propose a third-party solution that resolves the tromboning problem by replacing two international calls with two local calls. Besides huge saving in voice/data path, analytic modeling and simulation experiments indicate that our solution is effective in call setup signaling. Also, our solution does not need to modify the existing mobile telecommunications systems.

As internet telephony services based on Session initiation Protocol (SIP) enter the spotlight as marketable technology, many products based on SIPs have been developed and utilized for home and office telephony services. The call connection of an internet phone is classified into specific call connections and group call connections. Group call connections have a forking function which delivers the message to all of the group members. This function requires excessive message control for a call connection and creates heavy traffic in the network. In the internet cail system model. most of the call-setup messages are directed to the proxy server during a short time period. This heavy message load brings an unwanted delay in message processing and. as a result, call setup can not be made. To solve the delay problem, we simplified the analysis of the call-setup message in the proxy server, and processed the forking function distributed for the group call-setup message. In this thesis, a new system model to minimize the load is proposed and the subsequent implementation of this model demonstrates the performance improvement.

International audience; We consider a call center model with a callback option, which allows to transform an inbound call into an outbound one. A delayed call, with a long anticipated waiting time, receives the option to be called back. We assume a probabilistic customer reaction to the callback offer (option). The objective of the system manager is to characterize the optimal call scheduling that minimizes the expected waiting and abandonment costs. For the single-server case, we prove that non-idling is optimal. Using a Markov decision process approach, we prove for the two-server case that a threshold policy on the number of queued outbound calls is optimal. For the multi-server case, we numerically characterize a switching curve of the number of agents reserved for inbound calls. It is a function of the number of queued outbound calls, the number of busy agents and the identity of jobs in service. We also develop a Markov chain method to evaluate the system performance measures under the optimal policy. We next conduct a numerical study to examine the impact of the policy parameters on the system performance. We observe that the value of the callback offer is especially important for congested situations. It also appears that the benefits of a reservation policy are more apparent in large call centers, while they almost disappear in the extreme situations of light or heavy workloads. We moreover observe in most cases that the callback offer should be given upon arrival to any delayed call. However, if balking and abandonment are very high (which helps to reduce the workload) or if the overall treatment time spent to serve an outbound call is too large compared to that of an inbound one, there is a value in delaying the proposition of the callback offer.

S ince the calling range of current call treatment system is limited, resulting in staff retention in the waiting hall, noisy environment for treatment, work intensity of doctors and nurses, low treatment efficiency. A call treatment system based on MCU with STC89C52 MCU as the main control chip to control the ISD1760 voice broadcast, LED digital display and using RS232 serial port to form the call module .The smart phone is used as the receiving module to synchronously receive serial signals sent by WIFI module and voice information collected by FM chip .FPGA read the serial signals and convert them into VGA data, and then transmit it to the television display to form the display module. The patients who do not have smart phones or are not willing to install this APP can get the treatment process from the hospital's television screens or horns. Meanwhile, the patients who are on spot can synchronously check the treatment process through their smart phone. The results show that the system is stable and easy to operate. The call information can be conveyed to the waiting hall, and the Wi-Fi coverage area within a few hundred meters near the hospital.

A wireless call in a cellular network requires the cooperation of a wireless enabled device such as a cell phone and a dynamic set of base stations, antennas, and other network elements, with control of the call changing in response to changes in signal strength and the location of the mobile. The signaling between the network and mobile needed to manage the call generates a huge amount of signal strength data, some of which is seen only by the mobile placing the call and some of which is seen only by the network. This article describes some of the complexities of the signal strength data and provides a statistical model of the signals from the network to the wireless device that takes the time-dependent, spatial, and multivariate nature of the call into account. An approach to estimating the model parameters online as network data are collected, which would be useful for network monitoring, is also described and applied to a set of call data obtained from an active commercial CDMA (code division multiple...

Demand for value added services creates significant revenue opportunities for operators that decide to open their networks to external service providers. This process is practically feasible only if standardized interfaces to core network operations are utilized. This way the cost and complexity of the development and integration process is reduced, as well as the service administration overhead, which is distributed among service providers and does not burden the network operator. Most existing solutions for standardized access to core network operations follow the middleware paradigm that encapsulates network operations and provides open interfaces to outside service providers. The main concern about this approach has always been the performance impact of the additional architectural layers on top of the core network architecture. Different middleware technologies such as distributed object technologies and messaging middleware have been proposed and applied, all with varying performance results. The scope of this paper is to investigate and compare the performance impact inflicted by two such solutions encapsulating network call control functions. The first one is an OSA/Parlay based call control implementation with CORBA and RMI serving as underlying distributed object technologies and the second one is a JAIN call control implementation, again with an RMI underlying messaging framework. All performance measurements are taken in the context of a real-time ''call forwarding'' service translated from a corresponding VoIP SIP-based implementation.