• Journal of information and communication convergence engineering
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• v.8 no.5
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• pp.513-518
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• 2010

This paper proposes a medium access control protocol for integrated voice and data services in slotted CDMA systems. The proposed protocol, which is named as RCSSMA (Reservation Code and Status Sensing Multiple Access), adopts a code reservation and status sensing schemes. RCSSMA protocol gives higher access priority to the voice traffic than data traffic for reducing the packet dropping probability. The voice terminal reserves an available spreading code to transmit voice packets during a talkspurt, whereas the data terminal transmits a packet over one of the available spreading codes that are not reserved by the voice terminals. In this protocol, the voice packets never contend with the data packets. Packet dropping probability and average data packet transmission delay are analyzed using a Markov chain model.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.1067-1071
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• 2004

In this paper, the performance enhancement algorithm of channel allocation for voice and data transmission in cellular networks is proposed. The voice activity detection has been applied to dynamic channel allocation procedure to detect and separate the silence and speech among conversation periods. Hence a data user can use the silent period of an active voice channel to transmit its information. To control the selecting of channel allocation policies, the information of number of data in transmission waiting queue has been determined in order to accept the performance measurement. In the simulation results, the improvement of the performance shows via the quality of services, which are an average delay in queue, a blocking probability, and an impact of the proposed scheme is presented in the system.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.5A
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• pp.574-582
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• 2004

This paper shows the performance comparisons of several different rate scheduling schemes for non-real time data service over the uplink of burst switching-based direct sequencecode division multiple access (DS/CDMA) system to support the integrated voice/data service. The closed-form solution of optimal scheduling formulation, which minimizes average transmission delay when all of the active data users are transmitting simultaneously, is presented and mathematical analyses with other rate scheduling schemes, which provide efficiency criterion of transmission delay for rate scheduling schemes, are performed. Numerical results show the analyses explicitly.

• 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.

• Journal of Communications and Networks
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• v.14 no.5
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• pp.510-517
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• 2012

This paper analyzes the performance of voice-over-Internet protocol (VoIP) services in terms of the system throughput, the packet delay, and the signaling overhead in a mobile WiMAX system with a hybrid automatic repeat request (HARQ) mechanism. Furthermore, a queueing analytical model is developed with due consideration of adaptive modulation and coding, the signaling overhead, and the retransmissions of erroneous packets. The arrival process is modeled as the sum of the arrival rate at the initial transmission queue and the retransmission queue, respectively. The service rate is calculated by taking the HARQ retransmissions into consideration. This paper also evaluates the performance of VoIP services in a mobile WiMAX system with and without persistent allocation; persistent allocation is a technique used to reduce the signaling overhead for connections with a periodic traffic pattern and a relatively fixed payload. As shown in the simulation results, the HARQ mechanism increases the system throughput as well as the signaling overhead and the packet delay.

• Proceedings of the Korean Society of Computer Information Conference
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• 2010.07a
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• pp.173-179
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• 2010

Currently, the communication systems are progressing two ways as the wireless and multimedia and these need big transmission capacity then before. In these circumstance, communication services existed as two different service forms which have different rates and characteristics. For example Voice/Video Services accept some errors but transmit on realtime, but Date Services don't need to transmit on realtime but have to retransmit if these have only one bit error. In Voice/Date Integrated traffics, it has big throughput that realtime voice/video data which could have some errors if integrated traffic is increased rapidly have transmission priority, then Data traffics which delay is accepted is sent after that. In this paper, I introduce the calculation method for various traffic when voice/data mixed traffics is transmitted to asynchronous unslotted ALOHA CDMA system proposed and the result is presented. And We can easily theoretical analysis for the system traffic and changing traffic using proposed solution in this paper.

• The Journal of the Korea institute of electronic communication sciences
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• v.11 no.11
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• pp.1077-1082
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• 2016

IDS can be used as a countermeasure for malicious attacks which cause degrade of network transmission performance by disturbing of MANET routing function. In this paper, effects of IDS for transmission performance on MANET under grayhole attacks which has intrusion objects for a part of transmissions packets, some suggestion for effective IDS will be considered. Computer simulation based on NS-2 is used for performance analysis, performance is measured with VoIP(: Voice over Internet Protocol) as an application service. MOS(: Mean Opinion Score), CCR(: Call Connection Rate) and end-to-end delay is used for performance parameter as standard transmission quality factor for voice transmission.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.4 no.5
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• pp.691-708
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• 2010

An important issue of supporting multi-users with diverse quality-of-service (QoS) requirements over wireless networks is how to optimize the systematic scheduling by intelligently utilizing the available network resource while, at the same time, to meet each communication service QoS requirement. In this work, we study the problem of a variety of communication services over multi-media heterogeneous cognitive OFDM system. We first divide the communication services into two parts. Multimedia applications such as broadband voice transmission and real-time video streaming are very delay-sensitive (DS) and need guaranteed throughput. On the other side, services like file transmission and email service are relatively delay tolerant (DT) so varying-rate transmission is acceptable. Then, we formulate the scheduling as a convex optimization problem, and propose low complexity distributed solutions by jointly considering channel assignment, bit allocation, and power allocation. Unlike prior works that do not care computational complexity. Furthermore, we propose the FAASA (Fairness Assured Adaptive Sub-carrier Allocation) algorithm for both DS and DT users, which is a dynamic sub-carrier allocation algorithm in order to maximize throughput while taking into account fairness. We provide extensive simulation results which demonstrate the effectiveness of our proposed schemes.

• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.3
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• pp.275-280
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• 2010

In this paper, VoIP(Voice over Internet Protocol) transmission performance for MANET(Mobile Ad-hoc Networks) is improved and analyzed with packet aggregation scheme which is aggregating some of short length packets to one large packet and sending to networks. VoIP simulator based on NS(Network Simulator)-2 is implemented and used to measure performance of VoIP traffic transmission. In this simulation, VoIP traffics are generated with parameters of some codes such as G.711, G.729A, GSM.AMR and iBLC. MOS(Mean Opinion Score), end-to-end network delay, packet loss rate and transmission bandwidth are measured. Performance improvements of 98% for MOS, 6.4times for end-to-end network delay, 32times for packet loss rate is shown as simulation results. On the other hand, transmission bandwidth is increased about maximum 10%. Finally, VoIP implementation guide for the performance with packet aggregation is suggested.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.05a
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• pp.414-418
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• 2003

VoIP is a core technology used to transmit both voice and data in an integrated packet' form. Within this technology, SIP is the signaling protocol used for 'real time' call services; particularly those where H323 is used. Yet, when considering the needs of mobile users, it is essential we integrate VoIp within the mobile technology so the mobile host is able to receive the 'packet' transported and by, and connected to, any available internet-address. For all this to occur, we need to improve Network Delay by reducing transmission problems associated with mobile services. If we are to obtain an optimal service then we must reduce any network delays which may arise from joining Mobile IP and VoIp services. This paper, therefore, considers how, unlike previous research, these delays may be improved through the use of the signaling technology\ulcorner SIP. It also considers how this research may be introduced into current wired and wireless integrated services enabling them to use the IP 'packet'.