• 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 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.4B
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• pp.371-377
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• 2011

Users of the Information Age, IT usage patterns of the wired broadband information services and various forms of the same quality wireless multimedia services are required. Changes of these times the next-generation mobile communications (IMT-Advanced) has emerged as the necessity of developing its current voice and packet data communications on the move in the high-speed 100Mbps, 1Gbps in stationary and slow data transmission rates up to fixed-mobile convergence based on needed to provide ubiquitous service platform for the realization of IMT-Advanced is the time for preparation. In particular, 3-5GHz band, focused on mobile communications can be used to secure the necessary frequency band relocated and the existing crosstalk analysis methodology developed for the services rendered, and the frequency of such results to obtain new spectrum for IMT-Advanced for the country to secure the frequency characteristics and IMT-Advanced 3-5GHz band for the radio frequency of the characterization techniques necessary to develop a national wireless communication interference and frequency-based technology acquisition and management skills were identified.

• Journal of Korea Multimedia Society
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• v.14 no.1
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• pp.54-75
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• 2011

PoC(Push-to-talk Over Cellular) is an integrated technology of group voice calls, video calls and internet based multimedia services. If a PoC user can not participate in the PoC session for various reasons such as an emergency situation, lack of battery capacity, then the user can use the PoC Box which has a similar functionality to the MM Box in the MMS(Multimedia Messaging Service). The RTSP(Real-Time Streaming Protocol) method is recommended to be used when there is a transmission session between the PoC box and a terminal. Since the existing VOD service uses a wired network, the packet size of RTSP-based VOD service is huge, however, the PoC service has wireless communication environments which have general characteristics to be used in RTSP method. Packet loss in a wired communication environments is relatively less than that in wireless communication environment, therefore, a buffering latency occurs in PoC service due to a play-out delay which means an asynchronous play of audio & video contents. Those problems make a user to be difficult to find the information they want when the media contents are played-out. In this paper, the following techniques and methods were proposed and their performance and superiority were verified through testing: cross-over dual reception buffering technique, advance partition multi-reception buffering technique, and on-demand multi-reception buffering technique, which are designed for effective picking up of information in media content being transmitted in short amount of time using RTSP when a user searches for media, as well as for reduction in playback delay; and same-priority packetization transmission method and priority-based packetization transmission method, which are media data packetization methods for transmission. From the simulation of functional evaluation, we could find that the proposed multiple receiving buffering and packetizing methods are superior, with respect to the media retrieval inclination, to the existing single receiving buffering method by 6-9 points from the viewpoint of effectiveness and excellence. Among them, especially, on-demand multiple receiving buffering technology with same-priority packetization transmission method is able to manage the media search inclination promptly to the requests of users by showing superiority of 3-24 points above compared to other combination methods. In addition, users could find the information they want much quickly since large amount of informations are received in a focused media retrieval period within a short time.

• The KIPS Transactions:PartC
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• v.13C no.4 s.107
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• pp.471-480
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• 2006

Recently, multimedia communication services, such as video conferencing and voice over IP, have been rapidly spread. H.323 is an international standard that specifies the components, protocols and procedures that provide multimedia communication services of real-time audio, video, and data communications over packet networks, including IP based networks. H.323 is applied to many commercial services because it supports various network environments and has a good performance. But communication services based on H.323 may have some problem because of current network trouble or mis-implementation of H.323. The understanding of this problem is a critical issue because it improves the quality of service and is easy to service maintenance. In this paper, we implement the analysis system for H.323 protocol wihch includes H.245, H.225.0, RTP, RTCP, and so on. Tills system is able to capture, parse, and present the H.323 protocol in real-time. Through the operation test and performance evaluation, we prove that our system is a useful to analyze and understand the problems for communication services based on H.323.