• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.1
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• pp.7-16
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• 2012

In this paper we describe a VoIP transcript system which is able to record call conversation between counselor and customer in an IP contact center based on IP telephony environment. The transcript system, designed and implemented in this paper, uses packet sniffering to capture packets without imposing network overhead on overall system. It can decode H.323 and SIP which are used to setup call sessions in VoIP environment and captures voice data and record without any loss of contents. Implemented transcript system can be integrated with CTI system in that it can manage and record call more effectively. It is designed generically so that it is implemented both on Windows and Linux environment.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.5
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• pp.101-105
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• 2013

This paper addresses QoS functions in mobile backhaul network to accommodate the diverse traffics in cell site. The traffics assigned to the switching function in RAN system, such as Ethernet frame, IP packet, and ATM cell, are segmented, and then encapsulated to transfer then to the mobile backhaul network. ISP can converge all generation traffics, such as voice, HSPA, over all-IP RAN through standard pseudowire encapsulation. These can be enhanced with diverse QoS methods as well as comprehensive monitoring and diagnostic capabilities. Therefore in this paper, QoS functions under theses operations is simulated according to the encapsulation functions.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.8
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• pp.1696-1702
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• 2005

Real Time Transport Protocol (RTP) is the Internet standard protocol for transport of real time data audio/video IP Telephony, Multimedia Seivece. In case of 8kbps voice codec, the size of packet per data is 20bytes and become more large to minimal 40bytes with adding each layer's header in RTP/UDP/IP. To solve this problem, various header compression skill were suggested on point-to-point networks. But it compress even IP header and cannot be suitable to apply to end-to-end network Thus, We will renew header compression protocol to apply wired router-based network.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.12 no.7
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• pp.3275-3279
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• 2011

The matter of the calculation of the mutual access fee has become one of the hottest issues among service providers and attracted concerns from concerned regulatory authorities. It is essential to conclude a rational and systematic procedure for interconnection costs and charge between fixed and mobile networks. In this paper, I proposed the conversion factor scheme between circuit switched voice and packet switched data service in the domestic CDMA mobile system based on analysis of the rational GSM allocation method of common cost.

• The Transactions of the Korea Information Processing Society
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• v.6 no.2
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• pp.459-466
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• 1999

I broadband ISDN every packet will show different result if it would be processed according to its usage by the server. That is, normal data won't show big differences if they would be processed at normal speed. But it will improve the quality of service to process some kinds of data - for example real time video or voice type data or some data for a bid to by something through the internet - more fast than the normal type data. solution for this problem was suggested - priority packets. But the analyses of them are under way. Son in this paper a switching system for an output queuing model in a single server was assumed and some packets were given priorities and analysed. And correlation, simulating real life situation, was given too. These packets were analysed through three cases, first packets having no correlation, second packets having only correlation and finally packets having priority three cases, first packets having no correlation, second packets having only correlation and finally packets having priority and correlation. The result showed that correlation doesn't affect the mean delay time and the high priority packets have improved mean delay time regardless of the arrival rate. Those packets were assumed to be fixed-sized like ATM fixed-sized cell and the contention strategy was assumed to be short bus contention method for the output queue, and the mean delay length and the maximum 버퍼 length not to lose any packets were analysed.

• Journal of IKEEE
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• v.5 no.1 s.8
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• pp.8-15
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• 2001

This paper remarks IETF based requirements for IP/UDP/RTP header compression issued in 3GPP2 All IP Ad Hoc Meeting and protocol stacks of the next generation mobile station. All IP Network, for real time application such as Voice over IP (VoIP) multimedia services based on 3GPP2 3G cdma2000. Frames for various protocols expected in the All IP network Mobile Station (MS) are explained with several figures including the bit-for-bit notation of header format based on IETF draft of Robust Header Compression Working Group (ROHC). Especially, this paper includes problems of IS-707 Radio Link Protocol (RLP) for header compression which will be expected to modify in All IP network MS's medium access layer to accommodate real time packet data service[1]. And also, since PPP has also many problems in header compression and mobility aspects in MS protocol stacks for 3G cdma2000 packet data network based on Mobile IP (PN-4286)[2], we introduce the problem of solution for header compression of PPP. Finally. we suggest the guidelines for All IP network MS header compression about expected protocol stacks, radio resource efficiency and performance.

• The KIPS Transactions:PartC
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• v.10C no.7
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• pp.943-954
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• 2003

The mobile telecommunication system has been growing exponentially after 1990s due to the high population in a city and the growth of mobile user. In this time, the current mobile system mainly concentrates on the voice communication. However, in the next generation, mobile users want to get very diverse services via mobile terminal such as the Internet access, web access, multimedia communication, and etc. For this reason, the next generation system, such as the UMTS (Universal Mobile Telecommunication Services) system, has to support the packet data service and it will play the major role in the system. By the way, since the Web service is based on TCP, most of the Internet traffic TCP traffic. Therefore, efficient transmission of TCP traffic will take very important role in the performance of packet data service. There are many researches about improving TCP performance over wireless network. In those schemes, the UMTS system adapts the link layer retransmission scheme. However, there are rarely studies about the exact performance of the link layer retransmission scheme in the face of dynamic changes of wireless environment over the UMTS system. The dynamic changes of wireless environment, such as wireless bandwidth, can degrade TCP performance directly. So, in this paper, we simulate and analyze the TCP performance in the UMTS system with dynamic wireless environments. Then, we propose a simple scheme for minimizing TCP performance degradation. As a result of simulation, we can find that when wireless environment is changed dynamically, the probability of TCP timeout is increased, and the TCP performance is degraded very much. In this situation, the proposed simple scheme shows good performance. It saves wireless resources and reduces the degradation of TCP performance without large overhead of the base station.

• Journal of KIISE:Information Networking
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• v.29 no.4
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• pp.375-385
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• 2002

The IETF has defined the Intserv model and the RSVP signaling protocol to improve QoS capability for a set of newly emerging services including voice and video streams that require high transmission bandwidth and low delay. However, since the current Intserv model requires each router to maintain the states of each service flow, the complexity and the overhead for processing packets in each rioter drastically increase as the size of the network increases, giving rise to the scalability problem. This motivates our work; namely, we investigate and devise new control schemes to enhance the scalability of the Intesev model. To do this, we basically resort to the SCORE network model, extend it to fairly well adapt to the three services presented in the Intserv model, and devise schemes of the QoS scheduling, the admission control, and the edge and core node architectures. We also carry out the computer simulation by using ns-2 simulator to examine the performance of the proposed scheme in respects of the bandwidth allocation capability, the packet delay, and the packet delay variation. The results show that the proposed scheme meets the QoS requirements of the respective three services of Intserv model, thus we conclude that the proposed scheme enhances the scalability, while keeping the efficiency of the current Intserv model.

• Journal of KIISE:Computer Systems and Theory
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• v.26 no.8
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• pp.975-983
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• 1999

최근 네트워크의 속도가 빨라지고 멀티미디어 데이터를 다루기 위한 기술들이 개발됨에 따라 많은 멀티미디어 응용 프로그램들이 인터넷에 등장하고 있다. 그러나 이들 응용프로그램들은 수신자에게 전송되는 영상.음성의 품질이 낮기 때문에 기대만큼 빠르게 확산되지 못하고 있다. 영상.음성의 품질이 낮은 이유는 현재 인터넷이 실시간 응용프로그램이 요구하는 만큼 빠르고 신뢰성 있게 데이터를 전송할 수 없기 때문이다. 현재 인터넷의 내부구조를 바꾸지 않고 품질을 높이기 위해 많은 연구들이 진행되고 있는데 그 중 하나는 동적으로 변화하는 인터넷의 상태에 맞게 멀티캐스트 트래픽의 전송율을 조절하는 종단간의 흐름제어이다. 본 논문은 기존의 흐름제어 기법인 IVS와 RLM의 성능을 개선시키기 위한 두 가지 흐름제어 기법을 소개한다. IVS는 송신자가 주기적으로 측정된 네트워크 상태에 따라 전송율을 일정하게 조절한다. 송신자가 하나의 데이타 스트림을 생성하는 IVS와는 달리 RLM에서는 송신자가 계층적 코딩에 의하여 생성된 여러개의 데이타 스트림을 전송하고 각 수신자는 자신의 네트워크 상태에 맞게 데이타 스트림을 선택하는 기법이다. 그러나 IVS는 송신자가 전송율을 일정하게 증가시키고, RLM은 각자의 네트워크 상태를 고려하지 않고 임의의 시간에 하나 이상의 데이타 스트림을 받기 때문에 성능을 저하시킬 수 있다. 본 논문에서는 TCP-like IVS와 Adaptive RLM이라는 두 가지 새로운 기법을 소개한다. TCP-like IVS는 송신자가 전송율을 동적으로 결정하고, Adaptive RLM은 하나 이상의 데이타 스트림을 받기 위해 적당한 시간을 선택할 수 있다. 본 논문에서는 시뮬레이션을 통해 여러 가지 네트워크 구조에서 두 가지 방식이 기존의 방식에 비하여 더욱 높은 대역폭 이용율과 10~20% 정도 적은 패킷손실율을 이룬다는 것을 보여준다.Abstract Nowadays, many multimedia applications for the Internet are introduced as the network gets faster and many techniques manipulating multimedia data are developed. These multimedia applications, however, do not spread widely and are not fast as expected at their introduction time due to the poor quality of image and voice delivered at receivers. The poor quality is mainly attributed to that the current Internet can not carry data as fast and reliably as the real-time applications require. To improve the quality without modifying the internal structure of the current Internet, many researches are conducted. One of them is an end-to-end flow control of multicast traffic adapting the sending rate to the dynamically varying Internet state. This paper proposes two flow-control techniques which can improve the performance of the two conventional techniques; IVS and RLM. IVS statically adjusts the sending rate based on the network state periodically estimated. Differently from IVS in which a sender produces one single data stream, in RLM a sender transmits several data streams generated by the layered coding scheme and each receiver selects some data streams based on its own network state. The more data streams a receiver receives, the better quality of image or voice the receiver can produce. The two techniques, however, can degrade the performance since IVS increases its sending rate statically and RLM accepts one more data stream at arbitrary time regardless of the network state respectively. We introduce two new techniques called TCP-like IVS and Adaptive RLM; TCP-like IVS can determine the sending rate dynamically and Adaptive RLM can select the right time to add one more data stream. Our simulation experiments show that two techniques can achieve better utilization and less packet loss by 10-20% over various network topologies.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.1
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• pp.32-39
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• 2015

The voice communications with wireless backbone based networks are evolving into a packet switching VoIP systems. In those networks, a call processing scheme is required for management of subscribers and connection between them. A VoIP service scheme for those systems requires reliable subscriber management and connection establishment schemes, but the conventional call processing schemes based on the centralized server has lack of reliability. Thus, the mutual-backup architecture of SIP-servers is required to ensure efficient subscriber management and reliable VoIP call processing capability, and the synchronization and call processing schemes should be changed as the architecture is changed. In this paper, a mutual-backup architecture of SIP-servers is proposed for wireless backbone based networks. A message format for synchronization and information exchange between SIP servers is also proposed in the paper. This paper also proposes a FSM scheme for the fast call processing in unreliable networks to detect multiple servers at a time. The performance analysis results show that the mutual backup server architecture increases the call processing success rates than conventional centralized server architecture. Also, the FSM scheme provides the smaller call processing times than conventional SIP, and the time is not increased although the number of SIP servers in the networks is increased.