• International Journal of Railway
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• v.3 no.2
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• pp.39-45
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• 2010

The RF-CBTC (Radio Frequency-Communication Based Train Control) System is a communication system in railroad systems. The communication method of RF-CBTC system is the wireless between the wayside device and on-board device. The wayside device collects its location and speed from each train and transmits the distance from the forwarding train to the speed-limit position to it. The on-board device controlling device controls the speed optimum for the train. In the case of the RF-CBTC system used in Korea, transmission frequency is 2.4 [GHz]. It is the range of ISM(Industrial Scientific and Medical equipment) band and transmission of voice and data is performed by CDMA (Code Division Multiple Access) method. So noises are made in the AWGN (Additive White Gaussian Noise) and fading environment. Currently, the SNR (Signal to Noise Ratio) is about 20 [dB], so due to bit errors made by noises, transmission of reliable information to the train is not easy. Also, in the case that two tracks are put to a single direction, it is needed that two trains transmit reliable voice and data to a wayside device. But, by noises, it is not easy that just a train transmits reliable information. In this paper, we estimated the BER (Bit Error Rate) related to the SNR of voice and data transmission in the environment such as AWGN and fading from the RF-CBTC system using the CDMA method. Also, we supposed the SNR which is required to meet the BER standard for voice and data transmission. By increasing the processing gain that is a ratio of chip transmission to voice and data transmission, we made possible voice and data transmission from maximally two trains to a wayside device, and demonstrated it by using Matlab program.

• International journal of advanced smart convergence
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• v.1 no.2
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• pp.34-38
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• 2012

The recent increase in use of the IEEE 802.15.4 standard for wireless connectivity in personal area networks makes of it an important technology for low-cost low-power wireless personal area networks. Studies showed that voice communications over IEEE 802.15.4 networks is feasible by Guaranteed Time Slot (GTS) allocation; but there are some constraints to accommodate voice transmission beyond two hops due to the excessive transmission delay. In this paper, we propose a GTS allocation scheme for bidirectional voice traffic in IEEE 802.15.4 multihop networks with the goal of achieving fairness and optimization of resource allocation. The proposed scheme uses a greedy algorithm to allocate GTSs to devices for successful completion of voice transmission with efficient use of bandwidth while considering closest devices with another factor for starvation avoidance. We analyze and validate the proposed scheme in terms of fairness and resource optimization through numeral analysis.

• The KIPS Transactions:PartC
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• v.15C no.5
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• pp.367-374
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• 2008

To realize the voice privacy between CDMA mobile phone and PSTN terminal, the voice frames shall be transmitted transparently between the heterogeneous networks. For satisfying this requirement, we propose the method which transmits voice frames using the CDMA circuit data channel in real time. In this paper we analyze the causes of voice delay which occurs during voice transmission using circuit data channel. And in order to overcome this kind of delay, the technique controlling the TCP control flag and the variable audio block construction algorithm according to the vocoder output rate are proposed. As a result of experimenting by applying the proposed method, we confirmed that the transit delay was improved with about average 70%.

• Journal of Korean Society of Industrial and Systems Engineering
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• v.28 no.3
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• pp.133-141
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• 2005

The previous research has been mainly focused on individual technology of voice recognition, voice synthesis, translation, and bone transmission technical. Recently, commercial models have been produced using aforementioned technologies. In this research, a new automated translation support system concept has been proposed by combining established technology of bone transmission and wireless system. The proposed system has following three major components. First, the hybrid system consist of headset, bone transmission and other technologies will recognize user's voice. Second, computer recognized voice (using small server attached to the user) of the user will be converted into digital signal. Then it will be translated into other user's language by translation algorithm. Third, the translated language will be wirelessly transmitted to the other party. The transmitted signal will be converted into voice in the other party's computer using the hybrid system. This hybrid system will transmit the clear message regardless of the noise level in the environment or user's hearing ability. By using the network technology, communication between users can also be clearly transmitted despite the distance.

• The Journal of the Korea institute of electronic communication sciences
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• v.10 no.12
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• pp.1321-1328
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• 2015

A radio is a half-duplex voice communication method using the PTT(: Push To Talk), occupy a single line calls during transmission. As an interface between the telephone and the radio, UHF and VHF, for voice communication between the different heterogeneous devices, A device automatically switches between the two devices is required. Therefore, in accordance with the performance of the voice switching apparatus for detecting a voice to be transmitted from an input signal, loss of the audio signal to be transmitted is subjected to Significant influence. Conventional method has the problem responding to noise by setting the level through simple means of amplitude of input signal, in other words, the energy level of the input signal. This paper, by using the audio signal processing techniques, this discriminated what the voice is among the input signal and substantiated a device for the automatic voice transmission between heterogeneous devices. With this proposal, I was confirmed of improvement of performance in the automatic voice switching device, could perform loss-less transmission of voice between heterogeneous devices.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.17 no.3
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• pp.277-287
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• 1992

In this paper, we study the performance eualuations of the FDDI Network, by mathmeticlal analysis and simulation, in which the Integrated Voice/Data transmission system with voice traffic in synchronous mode and data traffic inasynchronous mode.For the mean waiting times of Voice/Data packet, we use two-state of Marcov models for voice traffic with talkspurt and silenci state, and the data traffic would traffic would transmit at the silence state of voice traffic. By the mean wating times, we analyze the relations between synchronous and asynchronous mode. As a result, using Sync/Async mode with voice and data, voice was not under influnece of data traffic. and in the same time,data can be tanaxmitted in a short waiting time, too.

• Journal of Internet Computing and Services
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• v.5 no.6
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• pp.31-43
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• 2004

Voice over Internet Protocol (VoIP) is currently a popular research topic as a real time voice packet transmission method. But current Internet environment do not guarantee the quality of voice when we take a side view of delay, jitter and loss. Up to now, many voice based evaluation algorithms have been used to measure speech quality of VoIP systems. However, these algorithms have the defects that their results are different according to voice samples and some algorithms can not take network environment for speech transmission path. The E-model can be used to solve the problems of these algorithms. In this paper. we introduce VoIP planning guidelines through the various analysis of E-model which can model impairments of network quality as well as VoIP equipment quality systematically, We, also, show the evaluation method and results of speech transmission quality.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.1
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• pp.259-266
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• 2013

Since the wireless communications have a limited transmission, the devices just around a master node can exchange data. Though Bluetooth and Zigbee support ad hoc, they are not appropriate for real-time voice communications. In this paper, we present a TDMA-based relay protocol for several users to communicate simultaneously. The proposed protocol can relay data or voice to other nodes in real-time by the multi-hop transmission method using TDMA. And the proposed protocol improves the network performance by allocating different frequencies to the slaves depending on the routing path scheduled by the routing table. NS-2 simulation shows that the performance of the proposed protocol is good in terms of the transmission delay and pecket loss probability in the real-time voice transmission.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2000.11a
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• pp.471-475
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• 2000

Voice and data are transmitted over a single packetized data communications network which is designed for data communications. The public switched telephone network for voice and the packet data network for data are merging into a single data network to get efficiency and to reduce operational cost. However, integrating voice and data transmission over a single data network is not easy because voice should be transmitted without delay but data should be transmitted without error. Advances in technology begin to overcome basic differences. Several integration methods in voice and data will be examined and reviewed here. Moreover, trends and problems on integration will be also discussed.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.12B
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• pp.2109-2119
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• 2000

In this paper, the Packet CDMA Reservation ALOHA protocol is proposed to support the multi-traffic services such as voice and videophone services with handoff calls, high-rate data and low-rate data services efficiently on the multi-rate transmission in uplink cellular systems. The frame structure, composed of the access slot and the transmission slot, and the proposed access permission probability based on the estimated number of contending users for each service are presented to reduce MAI. The assured priority to the voice and the videophone handoff calls is given through higher access permission probability. And through the proposed code assignment scheme, the voice service can be provided without the voice packet dropping probability in the CDMA/PRMA protocols. The code reservation is allowed to the voice and the videophone services. The low-rate data service uses the available codes during the silent periods of voice calls and the remaining codes in the codes assigned to the voice service to utilize codes efficiently. The high-rate data service uses the assigned codes to the high-rate data service and the remaining codes in the codes assigned to the videophone service. Using the Markov-chain subsystem model for each service including the handoff calls in uplink cellular systems, the steady-state performances are simulated and analyzed. After a round of tests for the examples, through the proposed code assignment scheme and the access permission probability, the Packet CDMA Reservation ALOHA protocol can guarantee the priority and the constant QoS for the handoff calls even at large number of contending users. Also, the data services are integrated efficiently on the multi-rate transmission.