Call Setup Time Research Articles

A data‐driven assessment of mobile operator service quality in Ghana

AbstractThe rapid proliferation of mobile services has increased the need for data‐driven oversight of service quality, yet deriving insights from regulator‐collected datasets remains challenging. This study demonstrates techniques to tap the rich potential of drive test measurement data for analytical regulatory and policy decision‐making. Focusing on leading operator MTN in Ghana, we analyzed 4 years of drive test data supplied by the telecom regulator for the capital city of Accra. Three key performance indicators were evaluated—coverage, call setup time, and speech quality. We assessed service quality trends through statistical summaries, data visualization, and machine learning modeling and predicted speech quality scores. Our analysis revealed deteriorating performance post‐2019 and found that the light gradient boosting machine algorithm provided the highest accuracy predictions of speech quality. Overall, this work showcases how regulators can capitalize on vast datasets using big data mining techniques to evaluate network conditions over time and geography, enhancing field measurements for oversight. Our approach and techniques provide a template for evidence‐based policy‐making to uphold consumer service quality as mobile networks evolve.

Performance Evaluation of Key Performance Indicators for UMTS Networks in Calabar, Nigeria

For appropriate regulation and quality assurance, continuous monitoring of cellular network services is necessary. A drive test measurement was carried out in Calabar, to investigate four UMTS networks, taking cognizance of Call Setup Success Rate (CSSR), Drop Call Rate (DCR), Handover Success Rate (HOSR), Call Setup Time (CST), Received Signal Code Power (RSCP) and Signal-to-Interference Ratio (Ec/Io). The collected data was analyzed and the obtained results was compared with the performance threshold levels of the Nigerian Communication Commission (NCC). The four networks had excellent CST, with MTN leading, followed by Airtel, then 9mobile and Globacom. For CSSR, Airtel had excellent performance, as there was no blocked call. This was followed by MTN and Globacom. 9mobile failed in CSSR as it was below the NCC recommended benchmark. There was no drop call in Airtel, MTN and 9mobile, but Globacom’s performance was below the recommended threshold for dropped calls. For HOSR, the four networks showed excellent performance as they were able to handover calls between cells, as recommended by NCC. Considering the RSCP, the four networks displayed excellent services as subscribers experienced good voice and data services. However, the four networks had poor coverage quality (Ec/Io), the worst, being displayed by MTN. This shows that the equipment capability of the networks is low. The network operators are advised to monitor their services often, so as to optimize their networks for a better Quality of Service (QoS).

A comparative assessment of GSM and UMTS Networks

It is common knowledge that the transition of mobile networks from one generation to another is basically for the improvement in the network’s Quality of Service (QoS). Bearing this in mind, we will assumme that the Universal Mobile Telecommunication System (UMTS) will outperform the Global System for Mobile Communication (GSM), hence, the motivation to conduct this study in Calabar, Nigeria, for four mobile networks; MTN, Airtel, Globacom and 9mobile. With the aid of a TEMS investigation software installed in a laptop, a measurement campaign was carried out and log files collected, with focus on Call Setup Success Rate (CSSR), Dropped Call Rate (DCR), Handover Success Rate (HOSR), Call Setup Time (CST), network coverage and network quality. The collected data was analyzed with the aid of a TEMS discovery software. The analyzed data for each Key Performance Indicator (KPI) was compared with the minimum benchmark of the telecommunications regulatory body, the Nigerian Communication Commission (NCC). Result reveal that there was no outright improvement in the QoS and there was fluctuation in the QoS provided by the network operators. We therefore conclude that the network operators, either did not make accurate planning before installing their base stations or do not optimize their networks frequently and this led to poor QoS in most cases.

A QOS ESTIMATION ALGORITHM FROM CALLER RINGTONE ANALYSIS IN GSM NETWORK

Call Setup Time (CST) is one of the key performance indicators (KPIs) that Mobile Network Providers (MNPs) are mostly appertain. It has been established that long CST usually severely affects the user experience. Owing to the limitations associated with gleaning the CST data from MNPs, this paper provides the development of QoS estimation algorithm from various CST parameters. The algorithm involves the determination of CST; Inter-Burst time; Intra-Burst time and Call duration in time domain. The caller Frequency content was also determined by the application of fast Fourier Transform before computing the Mean Square Error (MSE). The eventual QoS rating is done after the computation of the MSE from various individual parameters. Four hourly data consisting of 10 sets each were collected three times in a week for four weeks for each MNP’s for creating Caller Ringtone dataset and testing the developed algorithm. Performance analysis of the system in accurately determining: CST; Intraburst time; Interburst time and Call durations were carried out. Results obtained shows that the proposed technique accurately computes these parameters and maximum error obtained was to the value of 10%. Furthermore, the QoS obtained shows an error margin of less than 5 % was observed when the developed technique was compared to the ground truth. Thus, the proposed algorithm was able to compute the QoS using Caller Ringtone only, thus independent of MNP.

Robust Lightweight Privacy-Preserving and Session Scheme Interrogation for Fog Computing Systems

Session Initiation Protocol (SIP) is a communication protocol of multimedia systems providing IP signaling to initiate instant messaging. The rapid growth of wireless communication has gained momentum for the development of value-added services such as voice, data, and video ubiquitously. Continuous development has emerged in various application domains such as healthcare, surveillance, and automation that integrate the sources of multimedia to support real-time analysis. In mHealth, it has a dedicated multimedia information system (MIS) to support diagnosis, treatment, medicare, and report preparation. It may enable the user to collect the diagnostic information over smart devices. On the other hand, it may eliminate the geographical restriction between users and medical experts to manage medical emergencies. It uses a packet-switched network to cooperate with the wireless infrastructure to provide a standard interface that serves as a data transmission network. However, the communication interface such as real-time transport protocol is still challenging to authenticate the network services, key exchanges, and session blockage in the multimedia systems. Thus, this paper proposes a technique of session scheme interrogation (SSI) and robust lightweight privacy-preserving (RLPP) to resolve the major issues of LTE-A networks. In this paper, the session adaptive strategy is mathematically driven to adopt the session probing in multimedia clients that detects the session blocks proficiently to mitigate the session overheads. Besides, a real-time system has been designed and developed for the analysis of quality metrics such as call success rate, call setup time, end-to-end session delay, RTP utilization, bandwidth consumption, and flooding attack detection rate. The examination results prove that the proposed RLPP and SSI achieve better efficiencies than other existing schemes.

Simulator for Performance Evaluation of ASON/GMPLS Network

The hierarchical control plane network architecture of Automatically Switched Optical Network with utilization of Generalized Multi-Protocol Label Switching protocols is compliant to next generation networks requirements and can supply connections with required quality of service, even with incomplete domain information. Considering connection control, connection management and network management, the controllers of this architecture could perform the same operations on the transport plane resources and could support control functions for Software Defined Networking controllers, which are considered as the future networking solutions. Therefore, it is very important to determine factors that have an influence on service control and resource management. One of the tools for achieving this goal is computer simulation. The paper proposes a simulator for the hierarchical control plane networks in an open source OMNeT++ environment, in order to evaluate network architecture performance for different network structures and traffic parameters. To demonstrate simulator capabilities, examples of typical simulations performance results for Polish and European network structures are presented and discussed. Performance metrics which have been used to evaluate the performance of this architecture are: mean values of Call Set-up Time, Connection Set-up Time, Call Release Time, Connection Release Time, loss probability for inter-domain connection requests and loss probability for low and high priority requests. General remarks concerned with presented simulator are also provided.

IMPROVEMENT OF CALL SETUP TIME DURING CIRCUIT SWITCHED FALLBACK

In handover procedure, target cell would be prepared and UE will move on target cell based on the configuration sent by source eNodeB to User Equipment. Cell Reselection happens only when the user equipment (or Mobile Phone) is in Idle MODE, and the User Equipment needs to go a more appropriate cell. The main feature of CSFB Ps handover does not interrupt data transfer when switching from Long Term Evolution to UTRAN/GSM. After switching to 3G is already in connected mode it does not waste time to switch to connected mode. This feature let us improve call setup time when we make call in LTE.

A Cross-Layer Design for a Multihop, Self-Healing, and Self-Forming Tactical Network

In mission and time critical applications, bandwidth and delay optimizations are the key goals of communication systems. This paper presents a cross-layer framework design that reduces the call setup time, provides collision-free communication, and reuses the empty slots of Time Division Multiple Access (TDMA) protocol which otherwise causes low throughput and large delay. As number of communicating nodes in tactical networks is small as compared to commercial mobile ad hoc networks (MANETs), classical TDMA will yield huge number of empty slots and any Carrier Sense Multiple Access/Collision Detection (CSMA/CD) technique may cause more delay in some critical scenarios. Proposed methodology gives a Cross-Layer Architecture for Network (NET) Layer and Medium Access Control (MAC) Layer. Our design provides bandwidth efficient, collision-free communication to Software-Defined Radios (SDRs) in self-forming and self-healing tactical networks with low call setup time and multihop routing. For this purpose TDMA as MAC layer protocol and Ad Hoc On Demand Distance Vector (AODV) as Network Layer Routing Protocol are used. Our slot allocation (SA) algorithm, Cross-Layer TDMA (CL-TDMA), consists of control phase where AODV control packets are exchanged and data transfer phase where transmission of data and voice occurs. All active radios in vicinity gather information about communicating nodes based on the exchange of control packets by SDRs. Our algorithm then uses this information to help all active SDRs find slot(s) that will be used for collision-free transmission. A number of experiments are performed to establish improved performance of the proposed technique compared to other established techniques and protocols.

Traffic aware cell selection algorithm for Tetra Trunk based professional mobile radio

Load balancing and traffic management are the critical needs in cell selection decision for a better and seamless communication demands in professional mobile radios. For the cases where cell selection algorithms do not consider the traffic load, there may be call drops due to the congestion in networks or longer call setup times for the users. These undesired consequences can cause dramatic quality degradation especially for the emergency cases or public safety services. In this paper, we propose two algorithms for Tetra Trunk based professional mobile radios by considering both traffic load and received signal strength indication (RSSI) in order to reduce the significant delays while establishing transmissions. The proposed full set cell selection algorithm is prioritized to reduce the call setup time for the mobile users and the proposed reduced set cell selection algorithm is focused on minimizing the number of RSSI measurements which causes significant delay in practical professional mobile radio. We illustrate the performance results in terms of delay for Tetra Trunk based professional mobile radio.

ESAP: Efficient and secure authentication protocol for roaming user in mobile communication networks

The Global System for Mobile communication (GSM) network is proposed to mitigate the security problems and vulnerabilities observed in the mobile telecommunication system. However, the GSM network is vulnerable to different kinds of attacks such as redirection attack, impersonation attack and Man in-the Middle (MiTM) attack. The possibility of these attacks makes the wireless mobile system vulnerable to fraudulent access and eavesdropping. Different authentication protocols of GSM were proposed to overcome the drawbacks but many of them lead to network signalling overload and increases the call set-up time. In this paper, an efficient and secure authentication and key agreement protocol (ESAP-AKA) is proposed to overcome the flaws of existing authentication protocol for roaming users in the GSM network. The formal verification of the proposed protocol is presented by BAN logic and the security analysis is shown using the AVISPA tool. The security analysis shows that the proposed protocol avoids the different possible attacks on the communication network. The performance analysis based on the fluid flow mobility model shows that the proposed protocol reduces the communication overhead of the network by reducing a number of messages. On an average, the protocol reduces 60% of network signalling congestion overhead as compared with other existing GSM-AKA protocols. Moreover, the protocol not only removes the drawbacks of existing protocols but also accomplishes the needs of roaming users.

An intra–inter-cell device-to-device communication scheme to enhance 5G network throughput with delay modeling

As the number of mobile users is growing, so is the demand for more bandwidth. It becomes important that the required bandwidth and spectral resources do not scale with traffic in the next generation of wireless networks [i.e. fifth generation (5G)]. Device-to-device (D2D) communication underlaying cellular networks has been recognized as an essential technique in 5G networks. By applying definite principles and strategies, D2D communication not only increases the spectral and energy efficiency, but also enhances network throughput, network coverage and reduces delay. In this paper, we present an intra–inter-cell D2D communication scheme to enhance throughput of 5G networks. We study call setup delay of two developed communication scenarios and throughput gain comparing three systems. Firstly, we show the enhancements required in current cellular architectures to support inter-cell D2D communication. We develop protocols for two scenarios and demonstrate how architecture entities cooperate for the call setup between D2D users. We measure the overall call setup time for the developed protocols and derive a closed-form delay formula to estimate call setup time probability. Secondly, we perform simulations using a topology similar to that found in realistic urban environments to study throughput gains of the proposed intra–inter-cell D2D communication scheme. We compare three systems in terms of throughput: (1) pure cellular system (with cellular users only), (2) pure cellular system with intra-cell D2D users sharing the same cellular resource, and (3) pure cellular system with intra–inter-cell D2D users sharing the same cellular resource. Simulation results show that the proposed scheme substantially increases the network throughput and spectrum efficiency.

Kani: a QoS-aware hypervisor-level scheduler for cloud computing environments

Cloud computing environments (CCEs) are expected to deliver their services with qualities in service level agreements. On the other hand, they typically employ virtualization technology to consolidate multiple workloads on the same physical machine, thereby enhancing the overall utilization of physical resources. Most existing virtualization technologies are, however, unaware of their delivered quality of services (QoS). For example, the Xen hypervisor merely focuses on fair sharing of processor resources. We believe that CCEs have got married with traditional virtualization technologies without many traits in common. To bridge the gap between these two technologies, we have designed and implemented Kani, a QoS-aware hypervisor-level scheduler. Kani dynamically monitors the quality of delivered services to quantify the deviation between desired and delivered levels of QoS. Using this information, Kani determines how to allocate processor resources among running VMs so as to meet the expected QoS. Our evaluations of Kani scheduler prototype in Xen show that Kani outperforms the default Xen scheduler namely the Credit scheduler. For example, Kani reduces the average response time to requests to an Apache web server by up to $$93.6\,\%$$93.6%; improves its throughput by up to $$97.9\,\%$$97.9%; and mitigates the call setup time of an Asterisk media server by up to $$96.6\,\%$$96.6%.

A Dynamic-Identity Based Multimedia Server Client Authentication Scheme for Tele-Care Multimedia Medical Information System

Recently, several biometric and tele-care based user authentication schemes have been proposed to enhance the safe and security of the systems. In 2013, the authors, namely Das et al. improved the authentication scheme of Chang et al. to overcome the severe security flaws, such as failed to protect man-in-the-middle attack, failed to provide the reliable authentication and flaws in client login-phase. Besides, the authors evaluated the authentication in the simulation tool of AVISPA. In the same year, the authors, namely Khurram et al. proposed an improved the authentication scheme of Youngwa An to eradicate the security flaws, like impersonation attack, mutual authentication and user anonymity. The authors, like Das et al. and Khurram et al. have had their own strategies to mitigate the security flaws, though they are still not able to evaluate their schemes in the real time medical information systems. Besides, both authors fail to prove the privacy preservation to the user systems. Thus, this paper proposes a dynamic-identity based multimedia server client authentication scheme to resolve the major security threats of Das et al. and Khurran et al. We prove that our proposed scheme is secure and efficient in comparison with the authentication schemes, namely Youngwa An, Khurram et al., Das et al. and Chang et al. We also depict that our proposed scheme can offer the features like privacy preservation and service scalability reliably for the user systems. Eventually, we design an develop a real time testbed of multimedia medical information system to cross-examine the authentication schemes. In the cross-examination, our authentication scheme shows the most achievable results for the metrics like call setup time, signal congestion and bandwidth consumption in relation to the other authentication schemes, such as Youngwa An, Khurram et al., Das et al. and Chang et al.

IVisher: Real-Time Detection of Caller ID Spoofing

Voice phishing (vishing) uses social engineering, based on people's trust in telephone services, to trick people into divulging financial data or transferring money to a scammer. In a vishing attack, a scammer often modifies the telephone number that appears on the victim's phone to mislead the victim into believing that the phone call is coming from a trusted source, since people typically judge a caller's legitimacy by the displayed phone number. We propose a system named iVisher for detecting a concealed incoming number (that is, caller ID) in Session Initiation Protocol–based Voice-over-Internet Protocol initiated phone calls. Our results demonstrate that iVisher is capable of detecting a concealed caller ID without significantly impacting upon the overall call setup time.

An optimal sliding window size for maximising GSM network throughput

The throughput of a Global System for Mobile Communications (GSM) network is to be maximised for effectively and efficiently meeting the service requirements of the ever-increasing mobile subscriber base. The period of temporary storage of subscribers' profiles in the Visitor Location Register (VLR) at a Mobile Switching Centre (MSC) affects the throughput of a GSM network. This research study proposes a sliding window method for determining the storage period of profiles in the VLR, and the principle of optimisation for such a policy is to maximise the throughput and minimise the call set-up time in a GSM network.

Performance of CS Fallback from LTE to UMTS

Circuit Switched Fallback (CSFB) is the most commonly used method to support voice services over Long Term Evolution (LTE) networks today, as the deployment of IP Multimedia Subsystem (IMS) is still in its infancy. In this article, we discuss the performance of CSFB voice traffic redirection from LTE to UMTS using data from live commercial networks. The key factors impacting CSFB call setup delay are highlighted. Data from multiple cell geometries including stationary and mobility conditions are presented for Mobile Originated (MO) calls towards landlines, Mobile Terminated (MT) calls from landlines and Mobile-to-Mobile (M2M) CSFB calls. Analysis results in well optimized deployments show that on an average, MO/MT call-setup time for CSFB from LTE to UMTS is around 1 sec greater than legacy UMTS. However, the results can vary depending on the network configuration and the conditions of the measurement. To illustrate the optimization of CSFB performance in real networks, we also highlight the principal call set up optimization and implementation factors impacting CSFB call setup delay and success rates. This article demonstrates that in well optimized networks, CSFB to UMTS call setup success rates can be very close to those achieved in the legacy UMTS systems.

Hybrid Scheme with Adaptive Replication and Mobility Anchoring Point (MAP) in Wireless Networks

Replication strategy is mostly useful for replicating the mobile user's profile at selected locations from which some callers have higher calling rate. Location management scheme also include replication strategy to reduce the location lookup cost of called mobile user. It plays important role in personal communication services in order to reduce call setup cost, call setup time, and increases the QoS (quality of service). In prior replication schemes, called user's location profile is placed at some fixed and limited number of visitor location registers (VLRs) based on call history of the caller. In prior hybrid schemes location update and replication strategies are combined together. These prior hybrid schemes apply pointer forwarding strategy and replication. But these schemes have limitations like holding "loose" form of location profile and higher location update cost. We have proposed a new hybrid scheme which includes adaptive replication of updated user's location profile and mobility anchoring point (MAP) for location update. The main objective of the proposed scheme is to reduce call setup cost and call setup time. In order to achieve the objective, adaptive replication is applied at certain visitor location registers (VLRs) according to movement of caller. The replica is kept updated at MAP and replicated to neighbor VLRs as per movement of the caller. We have presented fluid-flow based analytical modeling which shows that proposed scheme outperforms the prior hybrid scheme.

Circuit Switch Fallback Improvement and the Smarter Way for Implementation

Operators, all over the world, search for the latest technologies and facilities to attract more customers. The LTE is the newest mobile technology, dedicated mainly for high-speed data transfer, and having all the chances to replace the existing legacy networks if it can provide voice services similar or even better from the existing. The delay on providing viable handsets supporting Single Radio Voice Call Continuity (SRVCC) solution forced many operators to use intermediate voice solutions, such as Circuit Switch Fallback (CSFB), not really elegant, in order to announce the support of voice services on their network, and overlook the fact of affecting badly their user's quality experience. We have proposed a smarter way for CSFB implementation, to clear all the confusions facing operators having still LTE network voiceless, and hesitating between the SRVCC solution, assumed to have viable handset on the coming months, or to the actual CSFB deployment. The first selection might give the precession in the media battle to other competitors, and the second selection might drain the cost resources reserved for voice over LTE project and consume more time depending on the width of the network. We have suggested also interesting recommendations to reduce the call setup time and improve the user quality experience, to overcome the main CSFB solution drawback.

User's Profile Replication Tree and On Demand Replica Update in Wireless Communication

Replication strategy is mostly useful for replicating the called mobile user's profile at selected locations where some caller mobile users have higher calling rate. It plays important role in personal communication services (PCS) in order to reduce call setup cost and call setup time. In prior replication schemes, replicated sites (visitor location registers) have no inter relationship. We have proposed a novel and innovative replication tree which is buildup from replicated VLRs (visitor location registers). VLRs are replicated based on effective replication probability of caller mobile user. Home location register (HLR) keeps update replica at root VLR in the replication tree and root VLR forwards update replica to a VLR in the replication tree on demand. The main objective of the proposed scheme is to reduce call setup time and call setup cost by enabling replication of called mobile user's location profile at certain visitor location registers (VLRs) where caller mobile user roams. We have presented analytical modeling which shows that setting inter-relationship among the replicated sites helps in further reducing the call setup cost.

The model of end-to-end call setup time calculation for Session Initiation Protocol

The model of end-to-end call setup time calculation for Session Initiation ProtocolEnd-to-end call setup delay is one of the most important grade of service (GoS) parameters for VoIP networks with Session Initiation Protocol (SIP). A typical subscriber wants to have a connection established as soon as possible. From the operator's perspective call setup time is also crucial because he needs to know how probability of successful packet transmission on SIP links is related to a call setup delay. Then he can make calculations how many resources are needed to guarantee such a transmission. In this paper we proposed the model for end-to-end call setup time calculation. The fundamentals of the model are to check all or almost all (highly probable) possible situations during the call setup. Formulas for calculation are included and results of numerical calculation are presented. Comparison against the values obtained from the simulation shows that it is possible to make calculations of end-to-end call setup time and prepare the discrete cumulative distribution function for a trapezoid model in reasonable time using the proposed model.