• 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%.

• Proceedings of the IEEK Conference
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• 2003.07d
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• pp.1629-1632
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• 2003

Generally an increasing number of recent applications have found TCP too limiting. There are some characteristics in the transmission of document and binary data which some transmission delay are tolerant but the content must completely be transferred. However voice signals are more sensitive with not some packet loss but some transmission delay. Therefore, Stream Control Transmission Protocol(SCTP) is proposed to minimize the delay and packet loss in the field of delivery of voice signal. SCTP is designed to transport PSTN signalling messages over IP networks, but is capable of broader applications. In this paper, the architecture of SCTP implementation is designed and some interface of SCTP software library which are implemented are specified.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.4
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• pp.1006-1025
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• 2012

Improving the quality of service in IP networks is a major challenge for real-time voice communications. In particular, packet arrival-delay variation, so-called "jitter," is one of the main factors that degrade the quality of voice in mobile devices with the voice-over Internet protocol (VoIP). To resolve this issue, a receiver-based enhanced timing recovery algorithm combined with active jitter estimation is proposed. The proposed algorithm copes with the effect of transmission jitter by expanding or compressing each packet according to the predicted network delay and variations. Additionally, the active network jitter estimation incorporates rapid detection of delay spikes and reacts to changes in network conditions. Extensive simulations have shown that the proposed algorithm delivers high voice quality by pursuing an optimal trade-off between average buffering delay and packet loss rate.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.13 no.2
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• pp.171-181
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• 1988

In this study, a packet voice/data terminal(PVDT) that services both voice and data in the packet-switched network is implemented. The software structure of the PVDT is designed according to the OSI 7 layer architecture. The discrimination of voice and data is made in the link layer. Voice packets have priority over data packets in order to minimize the transmission delay, and are serviced by a simple protocol so that the overhead arising form the retransmission of packets may be minimized. The hardware structure of the PVDT is divided into five modules; a master control module, a speech proessing module, a speech activity detection module, a telephone interface module, and an input/output interface module. In addition to the hardware implementation, the optimal reconstruction delay of voice packets to reduce the influence of delay variance is analyzed.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.2A
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• pp.187-194
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• 2004

We propose priority-based reservation code multiple access (P-RCMA) which can enhance voice traffic quality of the previous RCMA. The proposed protocol maintains two power levels and consider traffic characteristics in contending shared available codes to transmit packets. P-RCMA gives priority to the voice request packets rather than data packets by capture effect at the receiver part of base station. We show numerical results from EPA (equilibrium point analysis) analysis and simulation study in terms of voice packet dropping probability and average data packet transmission delay.

• The Journal of the Acoustical Society of Korea
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• v.18 no.6
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• pp.71-76
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• 1999

In digital cellular communication systems, tandem coding occurs in communications between mobile phones with different speech codecs, resulting in poor voice quality, high computational load, and long transmission delay. In this paper, voice packet conversion technique is proposed to solve the tandem coding problems, and packet conversion algorithm from 13kbps QCELP to 8kbps QCELP is developed. Simulations using various speech data show that the proposed packet conversion method produces voice quality which is equivalent to that by the conventional tandem coding method with shorter transmission delay using about 33% computational load.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.6
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• pp.29-35
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• 2013

The heterogeneous emergency communication radios is used at disaster management agencies of Korea to response activity in the event of disaster. The compensation method by communication connection time difference is necessary to seamless voice communication because radios have different communication method and delay. In this paper, we suggested solution for voice transmission chance and data loss problem.

• Journal of the Korean Institute of Telematics and Electronics
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• v.21 no.4
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• pp.24-30
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• 1984

Group delay distortion (phase distortion) is a characteristic which is of no account from a standpoint of voice transmission. But this distortion becomes the major source of distortion in wave form transmission such as data, FAX and others over the public switching telephone network (voice band transmission) so that it must be drastically studied. This paper makes analysis of group deray distortion of the telephone network, describs experimental and measuring results and refers also to the improvement of distortion for the purpose of opening the public switching telephone network to data transmission.

• 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.

• 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.