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With an ever-increasing number of complaints concerning “robo” and spam calls, the Federal Communications Commission (FCC) mandates service providers (SP) or internet telephone SP to filter or block unauthenticated or unverified calls. To U.S. mobile consumers, the “TRACED” law has passed enforcing the carriers to tag and, if needed, block spoofed numbers. The FCC requires major carriers to adopt and implement “STIR-SHAKEN,” a technology aimed at thwarting robocalls. The regulation imposes substantive fines, in the millions of dollars, for carriers that fail to implement. Alternatively, toll fraud hacks are on the rise, crippling telecommunication firms with revenue losses. Given the business problems with fraud and illegal scam calls, the technology aims at establishing a reliable end-to-end call authentication framework. This paper provides additional insight into technology trends to strategize new technology offerings, securing, and protecting telecommunications (telco) business operations.

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.

Smart devices such as smart phone play an important role in networked environments as connecting to other network devices and Internet Things. Smart phone combines the features of cell phone and a personal computer operating system. And it has become one of vital elements for modern people and society. However, smartphone has drawbacks like spam and security. Globally, people are exposed to spam call that they receive promotion calls from twice to dozens of times a day, spam call and voice phishing due to hacking and personal information leak in the internet environment. This phenomenon can cause personal stress, anxiety due to phone bank fraud, waste of time for receiving spam call, disturbance during work, criminal and the other serious social problems. In the present study, Humming Call system has been studied and developed to provide the measurement for evaluation and activation of service to prevent spam call in advance as a solution for the aforementioned problem. Hamming call is a service which receives information to help call through differentiation from existing service related to anti-spam system from DB (Internal DB) and Server. Then it provides the information for incoming call and outgoing call. To measure the actual performance by utilizing the system of this present study, DB has used CUBRID DBMS for 6 months and data stored in DB have been extracted through SQL Statement. The results indicate that this system prevents such problem for over 30%. If main function among various functions of this system is specialized and developed focusing on management of spam call from phone number, the system would perform more effectively.

As the number of IP-based telephony-like voice over IP (VoIP) users increases, interoperability of devices will become much more important to achieve smooth communications between arbitrary pairs of devices. To this end, detection and analysis of communication failures are key issues for validating interoperability. However, current failure (error) detections must be manually confirmed, which requires significant effort. Therefore, an automatic error-identification method is necessary. A straightforward approach can be considered that involves using a learning-based algorithm (e.g., k-nearest-neighbor-based classification) to identify errors; however, it is difficult to collect useful training data to achieve high identification accuracy. Since the number of conditions that we can prepare for in advance to collect training data is quite a bit less than that in a real environment, many errors may be misidentified when such cases are not included in the training data. To solve this problem, we propose a method which considers the characteristics of message sequences in a VoIP session control protocol (i.e., Session Initiation Protocol) by applying a similar approach of editing the distance between strings. Experiments using real environment data show that our method can improve the accuracy of identifying errors that are not included in the training data and enable administrators to collect new training data for unknown errors.

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.

Users have various human (natural) language needs, abilities, and preferences regarding spoken, written, and signed languages. This document adds new SDP media-level attributes so that when establishing interactive communication sessions ("calls"), it is possible to negotiate (communicate and match) the caller's language and media needs with the capabilities of the called party. This is especially important with emergency calls, where a call can be handled by a call taker capable of communicating with the user, or a translator or relay operator can be bridged into the call during setup, but this applies to non-emergency calls as well (as an example, when calling a company call center). This document describes the need and a solution using new Session Description Protocol (SDP) media attributes.

This paper proposes a Prioritized New Call Threshold Policy for Call Admission Control schemes. It is known that connection reattempts negatively affect the system performance. In currently proposed policy, the system's performance is improved by decreasing the number of connection reattempts. In order to demonstrate that the proposed scheme is better than the traditionally utilized ones in terms of various parameters, the simulations were utilized. It predicts the time that the mobile station will spend in the current cell, mobile station with different remaining time will have different number of channels available for them. More channels are available for the new calls with higher remaining time. Simulations were used in order to show that the proposed scheme outperforms the traditionally employed mechanisms in terms of various parameters.

Now-a-days, mobile phones are considered to be "always on, always connected" but mobile phone users are not always attentive and responsive to incoming phone calls. Incoming call notifications such as ringing at an inopportune moment (e.g., meeting) can cause interruptions for both the users and the surrounding people. In this paper, we present a system architecture for managing call interruptions according to individual's call response behavioral rules.

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.

Cellular phone networks are often paralyzed after a disaster. However, the infrastructure damage is difficult to be fixed within a day in the case of the disruptions caused by disasters. This paper addresses the challenges of proximity call communication in emergency and disaster areas. 3GPP Release 12 defines proximity service (ProSe) for public safety. When two users are making proximity call without infrastructure in a diaster, the on-going call is force-terminated as the users move out of the proximity range. In this paper, we propose a method to provide transparent service continuity via extending the ProSe call through a common ProSe neighbor without the need of interacting with the radio access network. Then we propose an analytical model to compute the call completion probability with the assist of common neighbor tunneling.

During base station subsystem (BSS) or radio access network (RAN) construction, the system capacity is planned according to the expected traffic volume in the coverage areas. When the traffic volume is within the planned capacity, the system can provide services properly. However, in certain situations, such as major events or in dense destinations, the traffic volume sometimes exceeds the planned capacity, leading to base stations overload. At times, it can happen due to unpredictable user’s behavior. If no measures are taken to protect the network, system performance may deteriorate noticeably. To avoid this, daily network performance KPI’s (Key Performance Indicator) observation is performed, but it is not enough, user’s habits are very important too. Especially when it comes to voice calls service, which is extremely important and widely used. Voice calls are one of the key services in today mobile business and require high quality of service. The focus of this research is on user habits of using voice call services in different locations. Main attention is paid to voice call duration measurement of various user groups. These user groups are composed from regular mobile network users who makes voice calls at six typical locations. The analysis includes voice calls duration distributions according to used technology for carrying a call, service payment type and other significant statistics. Collected data might be an important input for service providers or other parties, which are interested in humans communication habits. Comprehensible behavior of voice call users can help in selecting appropriate flow control mechanisms and ensure smooth call control management of circuit switched (CS) services.

A functional mapping between the Java Call Control (JCC) and Session Initiation Protocol (SIP) can help developers more easily create, deploy, and manage advanced telecom services.

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.

Service-oriented communication is a new trend in the industry to enable communication through a service-oriented architecture (SOA) and thereby package communications as services, which opens a new paradigm of Web service-based Voice Over Internet Protocol communication, which is extensible and can be easily integrated in end-to-end SOA solutions. In this paper, we proposed the communication components-based open multimedia conferencing Web service over IP networks and mainly focus on the session and media signalling for multimedia conferencing Web service. Details including the multimedia conferencing multiple parties call control state machines and session management, the state life cycle management, and the signal flows for multimedia conferencing scenarios, also including the Simple Object Access Protocol messages mapping to the asynchronous events. The session and media signalling management also provides the Web services interfaces for triggering conferencing session establishment and is further responsible for all necessary media signalling towards a proper media server. In addition, a prototype has been implemented, and its performance has been measured and analysed. Finally, we give the conclusions and future work. Copyright © 2011 John Wiley & Sons, Ltd.

While integrating telephony call control signaling into a desktop application is not a new concept, the emergence of web browser-based applications drives the need for feature parity between desktop and browser-based telephony call control applications. Desktop call control applications typically have the advantage of being able to use local protocol stacks or API libraries. A browser-based application does not have the same capabilities by default, especially when running on a constrained device. Traditional telephony call control APIs and architectures do not currently lend themselves for use with web browser-based applications without requiring some form of download which is not desirable. Is it possible to design a call control interface that can be consumed in browser-based applications natively without requiring additional downloads? Representational State Transfer (REST) is a resource oriented architectural style which, when used with HTTP as a uniform interface offers the possibility to use an existing defined set of interfaces to manipulate state transitions for resources. We propose that the REST architectural style can be used with an identified interface design methodology to define an interface for telephony call control capable of being consumed by browser-based applications. Furthermore by utilizing existing industry standards as an example, we suggest that existing standards can be extended to support browser-based integrations using REST.

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.

In a traditional intercom system that only allows place-to-place communication in a house, it is necessary to dial the extension number of a specific family member, or dial each extension number individually via the intercom broadcasting, in order to reach the desired person. Additional master stations controlled the intercom system are required, and it is load to maintain the stations for general users. In order to solve these problems, we propose the digital living network alliance (DLNA) compatible family intercom system (DFIS): the architecture to support user mobility. This architecture makes it possible to quickly reach a family member without the knowledge of the extension number that caters to a fixed location. In other words, it will be not necessary to dial the extension number of the location of the desired person. We introduce call control and phone handling under the subject of DFIS architecture and demonstrate that the proposed DLNA family intercom device and DLNA family intercom adaptor can work well.

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.

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.

A novel call admission control (CAC) scheme for an adaptive heterogeneous multimedia mobile network with multiple classes of calls is investigated here. Different classes of calls may have different bandwidth requirement, different request call holding time and different cell residence time. At any time, each cell of the network has the capability to provide service to at least a given number of calls for each class of calls. Upon the arrival (or completion or hand off) of a call, a bandwidth degrade (or upgrade) algorithm is applied. An arriving call to a cell, finding insufficient bandwidth available in this cell, may either be disconnected from the network or push another call out of the cell toward a neighboring cell with enough bandwidth. We first prove that the stationary distribution of the number of calls in the network has a product form and then show how to apply this result in deriving explicit expressions of handoff rates for each class of calls, in obtaining the disconnecting probabilities for each class of new and handoff calls, and in finding the grade of service of this mobile network

The Session Initiation Protocol (SIP) is widely used as a call control protocol for Voice over IP (VoIP), and indeed commercial implementations are readily available off-the-shelf. SIP supports flexible service provisioning not only through third parties, but also end-users. Laboratory experience shows that as these services are interworking they are subject to the feature interaction problem. Feature interactions may considerably delay service deployment and hence are a threat to rapid service provisioning. This paper investigates the feature interaction problem in SIP-based services and investigates the application of a pragmatic approach. This runtime approach does not require any detailed information about the services and hence can be applied in a competitive market. Furthermore, the technique is particularly strong in handling interactions between distributed services - a key characteristic of SIP-based services. Moreover, the approach is fully distributed without any centralised components, and includes detection and resolution of feature interactions.

This article investigates and analyzes the Handover of Work in a call center data previously collected from a telecommunication company in Thailand. The main objective of the study is to find the relationship values of the handover of work amongst the call center personnel in order to improve the efficiency of the handling with customers' receiving calls in total. To do this, we visualized the stored data based on the social network graphs with respect to handover of work metric. The main idea was to count the total number of times department J(in charge of specific received calls) performed an activity inbetween two activities performed by department I in such a way to indicate that work was subcontracted from department in to department J. Following the above-mentioned approach enabled us to easily and quickly track and trace the different departments (as well as the human resources dealing with the incoming calls) to attend and address the customers' receiving call in a more efficient, effective and timely manner.

In recent years, medical call centers have started using IP telephony services to minimize the overhead telecom expenses. However, the advent of Voice-Over-IP (VoIP) technology has also created a major discontinuity in telecommunication sector due to illegal VoIP or gray calls, causing a great impact on the voice market. This brought few challenges to the countries’ regulatory bodies. In this paper, we spell out one of the key challenges: in order to mitigate illegal VoIP calls, whether the regulatory body should allow IP telephony to be practiced for both domestic and international voice driven tele-medical consultation center operation combined or separately. We propose architecture and schemes for a medical call center. We also propose some guidelines and/or policies for both call center operator and the telecom regulatory authority. The proposed architecture and schemes are implemented in a pilot project basis in two phases and the test bed result is presented in this article.

Internet Telephony imposes Quality of Service (QoS) requirements in terms of delay, jitter and loss ratio to provide reliable, high-quality multimedia services. While the Internet has historically offered a best effort service, it is hard to satisfy the required QoS of multimedia applications. In order to provide a tight QoS guarantee to telephony services, priority must be offered to real-time flows. Dynamic resource negotiations are required at the beginning of sessions besides long term service contracts. Most of the researchers propose to integrate call signaling with resource negotiation in a sequential manner [G. Camarillo, W. Marshall, J. Rosenberg, Integration of Resource Management and SIP, IETF RFC 3312, October 2002; P. Goyal et al., Integration of call signaling and resource management for IP telephony. IEEE Network, 13(3) (1999) 24-32; H. Schulzrinne, J. Rosenberg, J. Lennox, Interaction of call setup and resource reservation protocols in Internet Telephony. Technical Report, Columbia University and Bell Laboratories, June, 1999. Available from: http:// www.cs.columbia.edu/~hgs/sip/drafts/resource.pdf]. This method can provide QoS at the highest level, but it results in high call setup delay and high probability of connection failure. Parallel integration is an alternative method that can mitigate these drawbacks. In this paper, we proposed a detail solution based on the parallel approach to establish resource negotiation simultaneously with call signaling. This approach can apply to different QoS models and mechanisms such as Integrated Services (IntServ), Differentiated Services (DiffServ) and Multi-Protocol Label Switching (MPLS). This paper presents the architecture and system design of the parallel signaling approach. Threshold quality control scheme is applied to guarantee that the service is above the acceptable level before a telephone session begins.

The need for policies to control calls is justified by the changing face of communications. It is argued that call control requires distinctive capabilities in a policy system. A specialised policy language called APPEL (ACCENT Project Policy Environment/Language) has therefore been developed for this purpose. However, the policy language is cleanly separated into a core plus specialisations for various application domains. The paper describes both the foundation and the call control ontologies. Sample policy examples are provided to illustrate use for call control. The paper also presents the policy system architecture in which the policy language is interpreted. The components of the policy system are described, particularly the policy server and the policy wizard.

System call monitoring is a technique for detecting and controlling compromised applications by checking at runtime that each system call conforms to a policy that specifies the program's normal behavior. Here, we introduce a new approach to implementing system call monitoring based on authenticated system calls. An authenticated system call is a system call augmented with extra arguments that specify the policy for that call, and a cryptographic message authentication code that guarantees the integrity of the policy and the system call arguments. This extra information is used by the kernel to verify the system call. The version of the application in which regular system calls have been replaced by authenticated calls is generated automatically by an installer program that reads the application binary, uses static analysis to generate policies, and then rewrites the binary with the authenticated calls. This paper presents the approach, describes a prototype implementation based on Linux and the PLTO binary rewriting system, and gives experimental results suggesting that the approach is effective in protecting against compromised applications at modest cost

Parlay Multi Media Call Control Services ( MMCCS ) Application Programming Interfaces ( API ) based on Session Initiation Protocol ( SIP ) is essential for the implementation of Parlay Call Control ( CC ) API . This paper first proposes an implementation model of Parlay MMCCS API based on SIP , then presents a mapping between SIP and MMCCS API as far as methods and parameters are concerned, illustrates corresponding relationships between different components of SIP and MMCCS API by means of an application collaboration diagram, and finally presents an application using Java codes as well as some SIP messages. The application shows that a majority of MMCCS API based on SIP can be implemented and therefore verifies our mapping between MMCCS API and SIP .

An efficient IP Multimedia Subsystem (IMS) online charging system is important for operator revenue generation. However, 3rd Generation Partnership Project (3GPP) specifications do not adequately address how online session charging shall be accomplished for IMS networks, especially how IMS network elements such as the application server and the serving call session control function (S-CSCF) interact with the online charging server (OCS). To remedy the shortcoming of 3GPP online charging standards, this paper presents a policy-based IMS call control system that converges the multiple application servers and the IMS gateway functionality into one IMS node for online session charging. The IMS session charging call control system employs a session charging trigger mechanism that enables a real-time report of session information and provides budget control for online quota monitoring. It also enables service logic for multiple applications to be executed in a single node to greatly reduce traffic access between the IMS call session control function (CSCF) and the OCS.

Voice over Internet protocol (VoIP) is a technology that enables the transmission of voice over an IP network. Recent years have witnessed heavy investment in this area in the commercial world. For VoIP to replace Public Switched Telephone Network (PSTN), it should provide voice quality comparable to circuit-switched PSTN networks. This paper addresses the mechanisms to guarantee VoIP quality of service (QoS). The focus is given to the call admission control, which blocks voice calls when the required resources are not available to guarantee the QoS for the call. We review call admission control approaches that can be applied to VoIP, and describe the advantages and disadvantages of each approach. In the second part of the paper, we present a measurement-based admission control scheme that achieves QoS in an efficient and scalable manner. The scheme uses voice traffic load measurements at each router link to compute link-level blocking policies for new call attempts. Then, these policies are translated into path-level blocking policies, which are applied to new call set-up requests. The performance of the scheme is presented for single and multiple-priority voice calls. Copyright © 2006 John Wiley & Sons, Ltd.

The need for policies to control calls is justified by the changing face of communications. An overview is given of a general architecture and language for policies. It is then shown how these are adapted for control of calls using the H.323 multimedia communications standard. Policy support for H.323 was created by extending an open source gatekeeper. The core policy language has been specialised to deal with call control in general, and for H.323 in particular. Examples are given of policies for H.323, illustrating how traditional features can be made to work more flexibly through use of policies. Examples are also provided of policies specific to H.323, and policies that can take advantage of other information such as the context of a call.

With an ever increasing penetration of IP technologies and the tremendous growth in wireless data traffic, the wireless industry is evolving the mobile core networks towards IP technology. The third Generation Partnership Project (3GPP) has specified an IP multimedia sub-system (IMS) in UMTS Release 5/6, which is adjunct to the UMTS packet-switched (PS) GPRS core network. This IP-based network provides full packet call control capabilities by using the text-based Session Initiation Protocol (SIP). Initial indications, as to the signalling delay associated with SIP messages, have concerned mobile operators about the viability of SIP services over the UMTS air interface. This article provides an insight into the UMTS system performance, focusing on selected UMTS SIP-based services. Typical services with real-time requirements such as voice as well as delay-sensitive and non-sensitive applications, such as real-time chat and instant messaging services are investigated. Furthermore, the paper discusses and analyses the requirements and possible solutions for improving efficiency of SIP usage in a wireless environment through signalling protocol message compression. Results of a performance evaluation of SIP signalling scenarios are presented in terms of time delay and message overload in the system. Results show that message compression can considerably reduce SIP message transmission time on the radio access network while core network delay contributions are found to be still high.

Interactive multimedia applications combine requirements of traditional telephony services and Internet applications. This requires call setup, call signaling, negotiation, routing, and network resources. However, in dynamic network environments resources are not protected unless the applications reserve them for the multimedia session. To provide quality of service (QoS) control, the subject of the interaction of call signaling and resource management have addressed resource reservation during the call setup phase. Usually, the rule is to use the traditional telephony-style call setup model (i.e. users are only alerted after all resources have been successfully reserved for the call). In the Internet, however, this may incur in long call setup delays and a high call blocking rate ('no QoS/no call'). In order to provide users with a practical and fast way to reserve resources during session setup, we investigate a different call setup model than that of traditional telephony. Our model allows users to be alerted while the resource reservation takes place, thus, taking advantage of parallel user answering delays and reservation delays. For this to occur efficiently, we use existing elements of the Session Initiation Protocol (SIP) to provide means for independent call setup and resource management transactions and flow negotiation. Based on experimental comparisons obtained in a multimedia call, we demonstrate that the proposed approach reduces the impact of the reservation delays, with maximum call setup delay improvements when we have approximate values of reservation and answering delays.

A CDMA mobile system consists of subsystems that transmit and receive call control messages and traffic messages through the CDMA Interconnection Network (CIN). To assure high throughput and short delay in the CIN, an efficient Medium Access Control (MAC) protocol should be provided. In this study, we propose a new MAC protocol for CDMA interconnection networks in order to provide real-time communications in CDMA mobile systems. Also, we evaluate the delay performance of the proposed MAC protocol and compare it with that of existing MAC protocol. Through numerical examples, we show that the proposed protocol provides better delay performance than the existing protocol.

Researchers propose a telephone availability and awareness network (TAAN) that could help businesses save time and money. The telephone is a cornerstone of modern life, giving nearly instant connections between friends, families, businesses and nations. Yet even modern systems consistently fail to enable users to make contact at the first attempt. A substantial proportion of business calls fail because the called party is unavailable and by the time a call is returned, the original caller is often unavailable. This imposes both a time and financial burden on the international business community, causing frustration and stress. Some in-house systems show when a colleague is engaged on a call, making it obvious when an internal call to that person would be pointless. This 'on-hook/off-hook' information could distributed more widely. Thus the idea of a telephone availability and awareness network was born. TAAN provides a solution to the problem of pointless telephone calls. The system is capable of indicating to a telephone user, in real time or substantially real time, the status or availability of a remote telephone line anywhere in the world. If necessary, it can function independently of the network being used for voice calls. The article outlines the concept behind TAAN.

Methodologies developed in the Internet Engineering Task Force (IETF) Internet Protocol Security (IPSEC) Working Group can be incorporated into asynchronous transfer mode (ATM) and frame relay (FR) signaling to provide message integrity and origin authentication. In turn, these mechanisms can provide a virtual private network (VPN) infrastructure with call control message integrity, origin verification, and transit network filtering. In this paper, we describe a set of control messages based on the IPSEC authentication header (AH) methodology that provide these security mechanisms for ATM and FR network switching equipment and signaling protocols.

The Parlay X specification describes a number of Web Services that will provide a simple interface to telephony and other systems. Other telephony interface initiatives have addressed the needs of complex and niche applications. The Parlay X initiative has set out to address the needs of the wider population of application developers. These application developers should be able to create telephony applications with little or no knowledge of the underlying network. Version 1 of the specification is complete and should be released to the public in the near future. This version covers third-party call control, SMS, multimedia messaging, payment, account management, user status and user location. This paper discusses the motivation for these new initiatives and presents an overview of the specifications and also gives an insight into the implementations available.

The session initiation protocol (SIP) is the main signaling protocol in the 3GPP IP multimedia subsystem (IMS). The SIP sessions in IMS have to be highly reliable. The developed fault-tolerant SIP call control concept includes state-sharing mechanism, failure-detection and fail-over management. The state-sharing mechanism representing the core entity in the overall system has been developed to meet the specific SIP functional features and requirements for reliability of SIP services. Theoretical analysis and measurements in a prototype implementation showed that TFTP outperforms FTP.

In this paper, a cyber space for multimedia conference is designed and implemented using gatekeeper and web service which satisfies H.323 standards. The implemented gatekeeper manages a registration and a call control for H.323 video conference terminals. The implemented gate-keeper can manage registered users efficiently with directory services using web service. Also, software maintenance of video conference system is established by service provider instead of users using control download on web. finally, we present the practical use of video conference through various applications of directory services.

This paper presents system call diversification as a method for protecting operating systems and rendering malicious programs ineffective. The idea is to change all the system call numbers in the kernel and in the applications that invoke these system calls. As a result, it becomes much more difficult for a harmful program to access resources of a computer since the new system call interface is not known by malware. The diversification of system call numbers is unique for each computer and the space of possible system call remappings is huge. Consecutively, one piece of malware no longer works on several computers and becomes incompatible with their environment. In this paper, we present three different models for system call diversification in Linux kernel. We also provide a detailed discussion on our implementation of one of these models.

In order to provide various multimedia services in a wireless network, the call admission control for wireless channels should be resolved at the time of call setup and handoff by moving mobile terminal. In this paper. we propose a reliable DCAC( Distributed Call Admission Control)scheme using virtual cluster concept. The proposed DCAC scheme considers the state of 1/sup st/ and 2/sup nd/ adjacent cells to provide a reliable call handling. The proposed scheme is analyzed by simulations and mathematical methods.l

The call admission control (CAC) belongs to the category of resource management. Since the radio spectrum is very scarce resource, CAC is one of the most important engineering issues for mobile communications. In this paper, we propose a CAC scheme for direct sequence code-division multiple-access cellular systems supporting mobile multimedia communications services. There are multiple call classes in multimedia services and the required signal-to-interference ratio (SIR) varies with call classes. Call admission decision in the proposed scheme is based on SIR measurement. We take account of the traffic asymmetry between uplink and downlink, which is the most important characteristic of multimedia traffic. In addition, the proposed scheme guarantees the priority of handoff call requests over new call requests. We evaluate the performance of the proposed scheme using Markov analysis. The performance measures which we focus on are the system throughput and the blocking probabilities of handoff calls and new calls. The outage probability of a call in progress is also calculated, which is the probability that the measured bit energy-to-noise density ratio of the call is smaller than the required value for maintaining adequate transmission quality. We present some numerical examples with practically meaningful parameter values and, as a result, show that the proposed CAC scheme can operate well in the mobile multimedia systems such as the International Mobile Telecommunications-2000 (IMT-2000) systems.

This letter examines third-party call control (3PCC) and its application to new services. The evolution of networks and services that has driven the need for 3PCC capabilities is examined along with some of the new service concepts that may make use of 3PCC. Also discussed is ongoing work within Lucent Technologies in the area of 3PCC and service concepts that make use of 3PCC.

Triggered by the deregulation of the telecommunications market, it is expected that the number of services deployed in the telecommunications network will increase dramatically in the near future. It is expected that these services will be interworking. However, as widely reported, interworking services will be subject to the service interaction problem. It is essential to cope with the problem in a developing multi service provider environment as it may substantially delay service deployment and hence form a serious obstacle to rapid service provisioning.In order to cope with the increasing number of services deployed in the network, previous work suggested the application of filtering approaches. This paper suggests such a technique. It is pragmatic in nature and does not require detailed service knowledge. Furthermore, it is shown how the approach can be incorporated into an overall comprehensive service interaction management process. Even though the approach is applied to call control aspects only, it could be extended to cover other areas.

The performance analysis of blended PSTN and IP call centers is likely to be in demand in the near future as the technology for these centers develops further. The authors did not find an analysis for a system of this type in the literature. The development of a user-friendly and portable tool based on their analysis methodology should be useful to organizations that have implemented, or are considering implementing, a blended call center. We have shown BCAT's wide range of uses. In the future, we plan to enhance BCAT to allow for skills-based routing to, for example, agents who can handle PSTN calls only, IP calls only, or both call types. This is a much more complicated queueing problem to model, but will provide increased flexibility to call center supervisors in terms of workforce management planning.

Software Managed Manycore (SMM) architectures -- in which each core has only a scratch pad memory (instead of caches), -- are a promising solution for scaling memory hierarchy to hundreds of cores. However, in these architectures, the code and data of the tasks mapped to the cores must be explicitly managed in the software by the compiler. State-of-the-art compiler techniques for SMM architectures require inter-procedural information and analysis. A call graph of the program does not have enough information, and Global CFG, i.e., combining all the control flow graphs of the program has too much information, and becomes too big. As a result, most new techniques have informally defined and used GCCFG (Global Call Control Flow Graph) -- a whole program representation which captures the control-flow as well as function call information in a succinct way -- to perform inter-procedural analysis. However, how to construct it has not been shown yet. We find that for several simple call and control flow graphs, constructing GCCFG is relatively straightforward, but there are several cases in common applications where unique graph transformation is needed in order to formally and correctly construct the GCCFG. This paper fills this gap, and develops graph transformations to allow the construction of GCCFG in (almost) all cases. Our experiments show that by using succinct representation (GCCFG) rather than elaborate representation (GlobalCFG), the compilation time of state-of-the-art code management technique [4] can be improved by an average of 5X, and that of stack management [20] can be improved by an average of 4X.

We present the architecture, design and experimental research prototype implementation of CitiTime, an open system architecture for the rapid development of advanced next-generation telephony services that overcomes some of the limitations of the current closed PSTN architecture and service model. CitiTime allows communication sessions to be set up over the PSTN, the Internet, or a combination of both. Services can be provided by multiple cooperating distributed service providers, some of whom may use third-party software components which can be “plugged in” or even dynamically downloaded from the network as needed. This allows advanced services to be deployed and delivered to users rapidly, a crucial requirement in the increasingly competitive telecommunications services marketplace. CitiTime is built upon an object-oriented call model called Citi Call Control (CCC) which we have defined as a small set of extensions to the standard Java Telephony API (JTAPI) call model. JTAPI is designed primarily for centralized, single provider, call center type applications. Our extensions provide support for multiple, distributed providers as well as advanced services. CCC hides details of underlying call-state management, protocols and hardware from applications. The CitiTime prototype software is currently operational in our laboratory. We briefly describe its current implementation as well as future work to address issues such as fault tolerance.

This paper examines current approaches to signalling and service interworking between Internet and public switched telephone network (PSTN). It notes that until now convergence between the two networks has for the most part taken place in the transport and signalling layers. Signalling interworking architectures cater for the specific class of telephony-like services and although they can accommodate the extension of the intelligent network's (IN) realm of control in the Internet, they do not provide a generic platform for service interworking. Through the adoption of the Parlay APIs, a way is foreseen for (a) consolidation of telephone service over both Internet and PSTN through the imposition of a uniform call control API while allowing the installed IN infrastructure to be used also for Internet telephony services and (b) for service interworking between telephony-like services and open distributed services in the Internet. The paper proposes a service architecture that can be used as a platform for Parlay-based service interworking while offsetting some drawbacks that the Parlay approach incurs.

Unfortunately, existing Telephone networks in India are unable to cope with huge demand due to limited bandwidth capacity. So the chances of successful Telephone call connection at the first attempt are rare. This can result in enormous frustration to the user to continue manual redialling until the call is established especially when important message is to be conveyed. A machine has been developed to save time of the busy user in trying to contact through telephone line. This paper discusses the features available in this machine and also explores many more exciting features which can be easily incorporated in this system.