• Journal of the Institute of Electronics Engineers of Korea TC
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• v.37 no.1
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• pp.32-43
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• 2000

In this paper, we investigate the existing medium access control (MAC) protocols to integrate the voice and data services in packet-based CDMA networks and furthermore, propose a new approach to circumvent the operational limits inherent in them. We propose the $P^2R$-CDMA (Prioritized Packet Reservation Code Division Multiple Access) protocol for the uplink in the synchronous multi-code CDMA system, which employs the centralized frame-based slot reservation along with the dynamic slot assignment in the base station using the QoS-oriented dynamic priority of individual terminal. The simulation results show that, as compared with the existing scheme based on the adaptive permission probability control (APC), the proposed approach can significantly improve the system capacity while guaranteeing the real-time requirement of voice service.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.8
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• pp.2090-2095
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• 1998

In this paper, we measured voice packet loss over the Korean Internet and analyzed percentage articulation by variation of the packet loss. To do this, we reviewed real-time transmission service based on RTP/UDP/IP and test method of the transmission quality. and implemented the real-time speech transmission system using GSM and UDP/IP. Monosyllable list has been chosen for the percentage articulation test, each voice packet has been coded and compressed by GSM and it has sequence number to measured packet loss and to recover out-of-order packets. In transmission results using seven router over the Korean Internet, we have show that loss rates reached 1.6% (unload), 22.5%(load) and loss rates after packet recovery by resequencing and FEC are from 9% to 35%. Finally, we have shown that percentage articulations by variation of the network traffic are Table 4.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.1B
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• pp.1-12
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• 2002

In the multimedia CDMA mobile communication service, it is required to support various applications, such as voice, video, file transfer, e-mail, and Internet access, with guaranteed QoS. In the mixed traffic environment ,which consists of voice, stream data, and packet data, we analyze the network where preemptive priority is granted to delay-intolerant voice service and a buffer is offered to delay-tolerant stream data service. And, for best-effort packet data service, the access control by transmission permission probability is applied to obtain prominent throughput. To analyze the multimedia CDMA mobile communication system, we build a 2-dimensional markov chain model about prioritized-voice and stream data services and accomplish numerical analysis in combination with packet data traffic based on residual capacity equation.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.14 no.2
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• pp.473-494
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• 2020

To provide a guaranteed Quality of Service (QoS) to real-time traffic in High-Speed Downlink Packet Access (HSDPA) core network, we proposed an enhanced mechanism. For an enhanced QoS, a Class-Based Low Latency Fair Queueing (CBLLFQ) packet scheduling algorithm is introduced in this work. Packet classification, metering, queuing, and scheduling using differentiated services (DiffServ) environment was the points in focus. To classify different types of real-time voice and multimedia traffic, the QoS provisioning mechanisms use different DiffServ code points (DSCP).The proposed algorithm is based on traffic classes which efficiently require the guarantee of services and specified level of fairness. In CBLLFQ, a mapping criterion and an efficient queuing mechanism for voice, video and other traffic in separate queues are used. It is proved, that the algorithm enhances the throughput and fairness along with a reduction in the delay and packet loss factors for smooth and worst traffic conditions. The results calculated through simulation show that the proposed calculations meet the QoS prerequisites efficiently.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.10B
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• pp.1852-1860
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• 1999

This paper studied performance of the AAL2 for variable rate real time services in ATM network with discrete-time simulation model. In this simulation, input parameters are packet fill delay for AAL2 PDU generation, guard time for ATM cell generation, burstness and number of channels. Though variation of the above mentioned parameters, we obtained end-to end delay variations and throughput, analyzed performance effect of the each parameter for voice packet service.

• ETRI Journal
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• v.28 no.3
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• pp.311-319
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• 2006

This paper presents a terminal-assisted frame-based packet reservation multiple access (TAF-PRMA) protocol, which optimizes random access control between heterogeneous traffic aiming at more efficient voice/data integrated services in dynamic reservation TDMA-based broadband access networks. In order to achieve a differentiated quality-of-service (QoS) guarantee for individual service plus maximal system resource utilization, TAF-PRMA independently controls the random access parameters such as the lengths of the access regions dedicated to respective service traffic and the corresponding permission probabilities, on a frame-by-frame basis. In addition, we have adopted a terminal-assisted random access mechanism where the voice terminal readjusts a global permission probability from the central controller in order to handle the 'fair access' issue resulting from distributed queuing problems inherent in the access network. Our extensive simulation results indicate that TAF-PRMA achieves significant improvements in terms of voice capacity, delay, and fairness over most of the existing medium access control (MAC) schemes for integrated services.

• Journal of information and communication convergence engineering
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• v.5 no.2
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• pp.88-92
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• 2007

In this paper, an SS-RRA protocol that is based on Code Division Multiple Access is proposed and analyzed under the integrated voice and data traffic load. The backward logical channels consist of slotted time division frames with multiple spreading codes per slot. The protocol uses a reservation mechanism for the voice traffic, and a random access scheme for the data traffic. A discrete-time, discrete-state Markov chain is used to evaluate the performance. The numerical results show that the performance can be significantly improved by a few distinct spreading codes.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.10
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• pp.1994-2005
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• 2015

Voice over IP (VoIP) is a technology for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks. Instead of being transmitted over a circuit-switched network, however, the digital information is packetized, and transmission occurs in the form of IP packets over a packet-switched network which consist of several layers of computers. VoIP Service that used the various techniques has many advantages such as a voice Service, multimedia and additional service with cheap cost and so on. But the various frauds arises using VoIP because VoIP has the existing vulnerabilities at the Internet and based on complex technologies, which in turn, involve different components, protocols, and interfaces. According to research results, during in 2012, 46 % of fraud calls being made in VoIP. The revenue loss is considerable by fraud call. Among we will analyze for Toll Bypass Fraud by the SIM Box that occurs mainly on the international call, and propose the measures that can detect. Typically, proposed solutions to detect Toll Bypass fraud used DPI(Deep Packet Inspection) based on a variety of detection methods that using the Signature or statistical information, but Fraudster has used a number of countermeasures to avoid it as well. Particularly a Fraudster used countermeasure that encrypt VoIP Call Setup/Termination of SIP Signal or voice and both. This paper proposes the solution that is identifying equipment of Toll Bypass fraud using those countermeasures. Through feature of Voice traffic analysis, to detect involved equipment, and those behavior analysis to identifying SIM Box or Service Sever of VoIP Service Providers.

• Convergence Security Journal
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• v.12 no.1
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• pp.35-42
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• 2012

Increase of data traffic and the advent of new real-time services require to change from the traditional TDM-based (Time Division Multiplexing) networks to the optical networks that soft and dynamic configuration. Voice and lease line services are main service area of the traditional TDM-based networks. This optical network became main infrastructure that offer many channel that can convey data, video, and voice. To provide high resilience against failures, Packet-optical networks must have an ability to maintain an acceptable level of service during network failures. Fast and resource optimized lightpath restoration strategies are urgent requirements for the near future Packet-optical networks with a Generalized Multi-Protocol Label Switching(GMPLS) control plane. The goal of this paper is to provide packet-optical network with a hierarchical multi-layer recovery in order to fast and coordinated restoration in packet-optical network/GMPLS, focusing on new implementation information. The proposed schemes do not need an extension of optical network signaling (routing) protocols for support.

• ETRI Journal
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• v.23 no.2
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• pp.52-60
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• 2001

In this paper, a medium access control protocol is proposed for integrated voice and data services in wireless local networks. Uplink channels for the proposed protocol are composed of time slots with multiple spreading codes per slot based on slotted code division multiple access (CDMA) systems. The proposed protocol uses spreading code sensing and reservation schemes. This protocol gives higher access priority to delay-sensitive voice traffic than to data traffic. The voice terminal reserves an available spreading code to transmit multiple voice packets during a talkspurt. On the other hand, the data terminal transmits a packet without making a reservation over one of the available spreading codes that are not used by voice terminals. In this protocol, voice packets do not come into collision with data packets. The numerical results show that this protocol can increase the system capacity for voice service by applying the reservation scheme. The performance for data traffic will decrease in the case of high voice traffic load because of its low access priority. But it shows that the data traffic performance can be increased in proportion to the number of spreading codes.