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Voice transmission over wireless edge networks is a conventional, cost-effective technology to transmit voice calls over 802.11 edge networks. In this type of service, Session Initiating Protocol (SIP) is responsible for initiating sessions. Research has shown that the call control algorithm that controls SIP sessions can also guarantee the quality of the medium during audio data transmission. In spite of decision-making in wireless access points and changing the parameters of the medium access sublayer (MAC), the existing algorithms are complicated and cannot adequately ensure service quality and efficient use of system resources. Therefore, the purpose of this study is to propose a new algorithm that enables a SIP server to decide to admit or reject an incoming call, in concise time. This algorithm considers the dynamic parameters of the network such as the location of node, channel busyness rate and real-time traffic percentage of the channel without changing the settings of MAC sublayer. The location of each system depending on the access point that transmits the call has a great impact on how the required service is provided. Implementation of this location-aware method on a real network testbed is indicative of remarkable superiority over the recently proposed methods in terms of service quality as measured by parameters, such as response time, call duration, loss-rate, and delay in real-time packets.

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.

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.

This chapter discusses productivity enhancements for the help desk through the implementation of computer telephony integration (CTI) technology. The decision to incorporate CTI technology into the planned state-of-the-art call center was mandated by call center management as a part of the move and necessary call center improvements. The present call center operation divides staff into focus areas, resulting in staff groups versed in the service requirements of one to two designated areas. This allows the development of representatives to meet specific customer needs. The CTI-specific project goal was to use all the advantages of third-party call control and associated middleware to reduce the amount of time spent handling each call. Calls route through the interactive voice response call flow via a script that supports common questions or, if necessary, forwards a call quickly to an appropriate representative to assist in the customer problem/solution. CTI implementation for the call center consisted of four separate stages.

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.

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.

The present research proposes a multi-attribute onboard call center intelligent queueing method applicable to the vehicle service field. In the previous studies concerning the call center queueing methods, the characteristics associated with the incoming call customers include the impatient waiting and the notification of waiting time, etc; however, the present paper takes into consideration a suite of additional customer attributes, i.e., emergency calls from intelligent onboard terminals, whether the call requests are from the area covered by the company’s 4S service, whether the call waiting time stays within a reasonable range, and whether there exists an necessity for engaging the third party service system. Also, priority classification is conducted based on the multiple attributes associated with the aforementioned customers. Besides, to optimize the service structure of the agent service representatives, the present study divides the agent service representatives into two groups, namely professional skill group and regular skill group. This allows the processing of call business to be carried out in two different modes. Correspondingly, preset processing time tables for the call business are established for regular and professional skill groups, respectively, so as to enable the service of agent representatives to better distinguish between regular business and complicated business, and conduct suitable processing accordingly. As a result, the processing efficiency is reinforced, the customers’ waiting time is reduced, and consequently the total processing time of agent service representatives is minimized. The research methodology adopted in the present manuscript is highly applicable to the design and management of call centers.

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.

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.

Caller-ID spoofing deceives the callee into believing a call is originating from another user. Spoofing has been strategically used in the now-pervasive telephone fraud, causing substantial monetary loss and sensitive data leakage. Unfortunately, caller-ID spoofing is feasible even when user authentication is in place. State-of-the-art solutions either exhibit high overhead or require extensive upgrades, and thus are unlikely to be deployed in the near future. In this paper, we seek an effective and efficient solution for 4G (and conceptually 5G) carrier networks to detect (and block) caller-ID spoofing. Specifically, we propose Nascent, Network-assisted caller ID authentication, to validate the caller-ID used during call setup which may not match the previously-authenticated ID. Nascent functionality is split between data-plane gateways and call control session functions. By leveraging existing communication interfaces between the two and authentication data already available at the gateways, Nascent only requires small, standard-compatible patches to the existing 4G infrastructure. We prototype and experimentally evaluate three variants of Nascent in traditional and Network Functions Virtualization (NFV) deployments. We demonstrate that Nascent significantly reduces overhead compared to the state-of-the-art, without sacrificing effectiveness.

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.

Quality Wireless has received customer complaints about long hold times at its call center. To address these complaints, it put into place certain process changes at its call center. After one month, the company will now decide whether improvement has taken place.To develop an understanding of process capability and how an improvement can be statistically validated. To understand the “check” phase of the plan-do-check-act cycle of Deming, using basic statistical principles.

Call centers have grown world-wide during the past decade. One of the most important aspects considered by call center managers is the optimization of its operators, which implies covering the highly variable demand and finding an efficient way to assign people to certain shifts in order to achieve a desirable service level and abandonment rate. Another challenge is determining which system setup is appropriate for the specific call center. Should we have a single-skill call center or multi-skill call center? If we do have the latter, how many multi-skill agents should we have on staff? In this case study, we generate and analyze discrete-event systems simulation-optimization models to test the behavior of a real-world call center under the actual configuration and under different levels of cross-training. The model results help call center managers by: 1) determining the optimal number of operators needed for different staff configurations in order to achieve the targets for service level and abandonment; 2) providing information about the trade-off between the key measurements in the call center; and 3) providing useful information about the number of operators needed and used for each hour of operation to estimate the number of four-hour shifts required to achieve the performance targets. Our experimental findings from this case study suggest that a bi-skill call center is economically better in the long-run compared to a full-skill or single-skill call center. This case study augments the call center body of knowledge by providing additional managerial insights for the operations management community.

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

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.

Among the various challenges of network and service monitoring, complexity is one of the hardest to overcome. The practical situation for cellular mobile networks and their services are especially troublesome, since there are three generations of mobile networks working together, and services are provided across these technologies. A very good example for such operations and maintenance nightmare is the Voice over LTE (VoLTE) service, currently under deployment worldwide. In theory it merely requires the interworking of the EPC (Evolved Packet Core) of LTE and the IMS (IP Multimedia Subsystem) for call control. In reality the 2G and 3G mobile cores get involved as well, when 4G coverage is not satisfactory, and SRVCC (Single Radio Voice Call Continuity) procedures kick in. Complete monitoring, including cross- protocol data analysis of VoLTE control is a key issue that currently in the operators' interest. In this paper we introduce the problem, and provide the general solution as well as special insights for cross-protocol analysis of the VoLTE service. Such analysis results can provide a basis for the next generation of mobile core monitoring, based on supervised learning.

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.

Given a set of request calls with different demands and penalty costs, the prize-collecting call control (PCCC) problem is to minimize the sum of the maximum load on the edges and the total penalty cost of the rejected calls. In this paper, we prove that the PCCC problem on weighted lines is NP -hard even for special cases, and design a 1.582-approximation algorithm using a randomized rounding technique. In addition, we consider some special cases of the PCCC problem on weighted lines and rings.

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.

The majority of call admission control schemes proposed for wireless cellular networks do not take new call redials into account. However, connection reattempts are known to have a negative impact on the system performance because they generate more load. In this paper, we shall investigate how some popular call admission control schemes perform under more realistic conditions.

In this chapter, we present how routing works in the global switched telephone newtork (GSTN). We disucss E.164 addressing for phone devices. We also discusses network and carrier addressing that are used for transfering a call between different providers. We also disuss SS7, the signalling network, and how this is used with circuits for the call network. Finally, we discuss local number portability and its impact on call routing.