• Proceedings of the IEEK Conference
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• 2003.07a
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• pp.61-64
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• 2003

GPRS is designed for transmitting packet data and supposed to take its radio resource form the pool of channels unused by GSM voice services. In this paper, The GPRS and GSM circuit switched services share the same radio resource. Whenever a channel is not used by circuit switched services, it may be utilized by GPRS. In this paper, the main aim is performance and comparison resource management policies with channel de-allocation in GPRS network. Three resource management policies is voice priority, R-reservation, dynamic reservation.

• Journal of Communications and Networks
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• v.11 no.1
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• pp.36-41
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• 2009

The IEEE 802.16 wireless metropolitan area network (WMAN) standard is designed to correct expensive communication costs in CDMA-based mobile communication systems and limited coverage problems in wireless LAN systems. Thus, the IEEE 802.16e standard can provide mobile high-speed packet access between mobile stations and the Internet service provider through the base station with cheap communication fees. To efficiently accommodate voice over IP (VoIP) services in IEEE 802.16 systems, an uplink quality of service packet-scheduling scheme is proposed, and its performance is evaluated with an NS-2 network simulator in this paper. Numerical results show that this proposed scheme can increase the system capacity by 100% more than in the unsolicited rand service (UGS) scheme and 30% more than the extended real-time polling service (ertPS) scheme, respectively.

• Speech Sciences
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• v.11 no.4
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• pp.67-73
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• 2004

Packet loss, jitter, and delay in the Internet are caused mainly by the shortage of network bandwidth. It is due to queuing and routing process in the intermediate nodes of the packet network. In the Internet whose bandwidth is changing very rapidly in time depending on the number of users and data traffic, controlling the peak transmission bit-rate of a VoIP. system depending on the channel condition could be very helpful for making use of the available network bandwidth. Adapting packet size to the channel condition can reduce packet loss to improve the speech quality. It has been shown in [1] that a VoIP system with an AMR speech codec provides better speech quality than VoIP systems with fixed rate speech codecs. With the adaptive codec mode assignment. algorithm proposed in [1], in this paper, we performed the voice transmission experiments using the wireless LAN through the real Internet environment. Experimental results are analyzed and discussed with our findings.

• Journal of Advanced Marine Engineering and Technology
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• v.32 no.5
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• pp.768-774
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• 2008

In many areas of speech signal processing such as automatic speech recognition and packet based voice communication technique, VAD (voice activity detection) plays an important role in the performance of the overall system. In this paper, we present a new feature parameter for VAD which is the product of energy of the signal and the difference of two types of entropies. For this end, we first define a Mel filter-bank based entropy and calculate its difference from the conventional entropy in frequency domain. The difference is then multiplied by the spectral energy of the signal to yield the final feature parameter which we call PEED (product of energy and entropy difference). Through experiments. we could verify that the proposed VAD parameter is more efficient than the conventional spectral entropy based parameter in various SNRs and noisy environments.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.14 no.4
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• pp.360-368
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• 1989

In this paper, the performance of an integrated voice/data cut-through switching network is studied. We first derive cut-through probabilities of voice and data packets at intermediate nodes. Then, the Laplace transform for the network delay is obtained. According to numerical results, the performance of cut-through switching is superior to that of packet switching for integrated voice/data networks.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.8B
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• pp.683-690
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• 2003

In this paper we present a theoretical framework for the network design with delay QoS guarantee to a voice at the packet level. Especially, we propose a method for estimating the bandwidth at the ingress edge routers accommodating the voice connections and data sessions in the next-generation If network. First, we describe network architecture for VoIP (Voice over IP) services in the NGN (Next Generation Network). After that, we propose a procedure for dimensioning the bandwidth at the output port of a router that accommodates voice and data traffic using the non-preemptive queuing system with strict priority service scheme. Via numerical experiments we illustrate the implication of the proposition.

• International journal of advanced smart convergence
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• v.9 no.4
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• pp.209-214
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• 2020

VoIP technology has been widely used for exchanging voice or image data through IP networks. VoIP technology, often called Internet Telephony, sends and receives voice data over the RTP protocol during the session. However, there is an exposition risk in the voice data in VoIP using the RTP protocol, where the RTP protocol does not have a specification for encryption of the original data. We implement programs that can extract meaningful information from the user's dialogue. The meaningful information means the information that the program user wants to obtain. In order to do that, our implementation has two parts. One is the client part, which inputs the keyword of the information that the user wants to obtain, and the other is the server part, which sniffs and performs the speech recognition process. We use the Google Speech API from Google Cloud, which uses machine learning in the speech recognition process. Finally, we discuss the usability and the limitations of the implementation with the example.

• 한국정보통신설비학회:학술대회논문집
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• 2009.08a
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• pp.275-280
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• 2009

Femtocells are viewed as a promising option for mobile operators to improve coverage and provide high-data-rate services in a cost-effective manner Femtocells can be used to serve indoor users, resulting in a powerful solution for ubiquitous indoor and outdoor coverage. TThe frequency accuracy and phase alignment is necessary for ensuring the quality of service (QoS) forapplications such as voice, real-time video, wireless hand-off, and data over a converged access medium at the femtocell. But, the GPS has some problem to be used at the femtocell, because it is difficult to set-up, depends on the satellite condition, and very expensive. The IEEE 1588 specification provides a low-cost means for clock synchronisation over a broadband Internet connection. The Time of Packet (ToP) specified in IEEE 1588 is able to synchronize distributed clocks with an accuracy of less than one microsecond in packet networks. However, the timing synchronization over packet switched networks is a difficult task because packet networks introduce large and highly variable packet delays. This paper proposes an enhanced filter algorithm to reduce ths packet delay variation effects and maintain ToP slave clock synchronization performance. The results are presented to demonstrate in the intra-networks and show the improved performance case when the efficient ToP filter algorithm is applied.

• Journal of Information Processing Systems
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• v.2 no.2
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• pp.107-113
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• 2006

It is essential to guarantee a handoff dropping probability below a predetermined threshold for wireless mobile networks. Previous studies have proposed admission control policies for integrated voice/data traffic in wireless mobile networks. However, since QoS has been considered only in terms of CDP (Call Dropping Probability), the result has been a serious CBP (Call Blocking Probability) unfairness problem between voice and data traffic. In this paper, we suggest a new admission control policy that treats integrated voice and data traffic fairly while maintaining the CDP constraint. For underprivileged data traffic, which requires more bandwidth units than voice traffic, the packet is placed in a queue when there are no available resources in the base station, instead of being immediately rejected. Furthermore, we have adapted the biased coin method concept to adjust unfairness in terms of CBP. We analyzed the system model of a cell using both a two-dimensional continuous-time Markov chain and the Gauss-Seidel method. Numerical results demonstrate that our CAC (Call Admission Control) scheme successfully achieves CBP fairness for voice and data traffic.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.6
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• pp.17-22
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• 2013

Heterogeneous radios are used in disaster management agencies for a variety of reasons though the radio must have the same radio frequency and protocol for voice communication. For this reason, the variety of heterogeneous radio voice connection methods have been studied but these are simple analog voice line cross connection or partial networked based on digitalization. In this paper, we suggest the method of voice packet transmission method based on All-IP per radio through IP network using SIP/RTP for scalability and openness and developed a prototype of the proposed method was verified.