• Proceedings of the Korea Contents Association Conference
• /
• 2004.05a
• /
• pp.342-347
• /
• 2004

The VoIP(Voice over Internet Protocol) technology which is able to use a voice service through internet is more cheaper then existing telephone charges, and is easily accept the various of multimedia services from internet. Previous connection method of VoIP used H.323 protocol, but it is very complex to connection establishment. so, the SIP(Session Initiation Protocol) protocol that propose in SIP-Working Group Is in use recently. Therefore, we need new interworking methodology between H.323 and SW products. In this thesis, the progress interworking method between H.323 and SIP are propose, then interpret unnecessary packet delay for call setup and improved feature of message exchange.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.6 no.6
• /
• pp.931-936
• /
• 2002

By emphasize and enhance security in encryption of same viewdata in this research, constructed video conference system that apply real time video encryption techniques that confidentiality and integrity aspect are emphasized than video conference system that depend on existent user certification techniques. Sender's image transmits Vernam's encryption process to listener by packet. Listener can display image through image's decipher process and uncompress after unite receiving transmited image. This time, solved transmission speed delay problem by video's real time encryption using Vernam's algorithm that encryption and the decipher speed are fast. Simplification problem of Vernam algorithm wished to solve Session Key that use security Key that is used encipherment and decipher in channel between Client and Server using RSA algorithm.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.10 no.12
• /
• pp.5191-5211
• /
• 2016

It is essential to satisfy class-specific QoS constraints to provide broadband services for new generation networks. The present study proposes a QoS-driven multicast scheme for wireless networks in which the transmission rate and end-to-end delay are assumed to be bounded during a multiple multicast session. A distributed algorithm was used to identify a cost-efficient sub-graph between the source and destination which can satisfy QoS constraints of a multicast session. The model was then modified as to be applied for wireless networks in which satisfying interference constraints is the main challenge. A discrete power control scheme was also applied for the QoS-aware multicast model to accommodate the effect of transmission power level based on link capacity requirements. We also proposed random power allocation (RPA) and gradient power allocation (GPA) algorithms to efficient resource distribution each of which has different time complexity and optimality levels. Experimental results confirm that the proposed power allocation techniques decrease the number of unavailable links between intermediate nodes in the sub-graph and considerably increase the chance of finding an optimal solution.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.25 no.10B
• /
• pp.1665-1674
• /
• 2000

This paper treats the interworking of LAN-based networks like TCP over the ATM protocol stack in an ATM multicast session. Multicast connection will cause CIP since multicast group members form a connection tree by some tree methods and share the connected tree. The paper solve the CIP problem through a block-by-block transmission using ABT/IT method. ABT/IT RM cell is modified and block scheduling algorithm considering the traffic types is applied to each ATM switch using the enhanced RM cell. Block scheduling algorithm will avoid the indiscriminate discard of UBR traffic when congestion occurs and it can provide an efficient and fair service. The paper builds a block scheduler system and suggests the block scheduling algorithm for a multicast session in an ATM switch. UBR traffics arriving at the switch trough each VC is classified by the traffic type and stored at class buffer and thereafter indisciminately transmitted. When block scheduling algorithm is applied it will improve the UBR traffic performance such as end-to-end delay cell block loss ration etc. This paper evaluated the performance of block scheduling algorithm through the simulation using the C language and data structure.

• Journal of Communications and Networks
• /
• v.14 no.5
• /
• pp.578-587
• /
• 2012

One of the key elements in the emerging, packet-based long term evolution (LTE) cellular systems is the deployment of multiple femtocells for the improvement of coverage and data rate. However, arbitrary overlaps in the coverage of these femtocells make the handover operation more complex and challenging. As the existing handover strategy of LTE systems considers only carrier to interference plus noise ratio (CINR), it often suffers from resource constraints in the target femtocell, thereby leading to handover failure. In this paper, we propose a new efficient, multi-objective handover solution for LTE cellular systems. The proposed solution considers multiple parameters like signal strength and available bandwidth in the selection of the optimal target cell. This results in a significant increase in the handover success rate, thereby reducing the blocking of handover and new sessions. The overall handover process is modeled and analyzed by a three-dimensional Markov chain. The analytical results for the major performance metrics closely resemble the simulation results. The simulation results show that the proposed multi-objective handover offers considerable improvement in the session blocking rates, session queuing delay, handover latency, and goodput during handover.

• Journal of the Korea Institute of Military Science and Technology
• /
• v.15 no.6
• /
• pp.770-778
• /
• 2012

In this paper, we propose a multi-path construction scheme to improve the survivability of a multicast session in military hybrid networks. A military hybrid network consists of a static backbone network and multiple mobile stub networks where some nodes are frequently susceptible to be disconnected due to link failure and node mobility. To improve the survivability of multicast sessions, we propose a construction scheme of ${\kappa}$ redundant multi-paths to each receiver. In order to take account of different characteristics of static and mobile networks, we propose quite different multi-path setup approaches for the backbone and stub networks, respectively, and combine them at the boundary point called gateway. We prove that our proposed scheme ensures that each receiver of a multicast session has ${\kappa}$ redundant paths to the common source. Through simulations, we evaluate the performance of the proposed schemes from three aspects : network survivability, recovery cost, and end-to-end delay.

• Journal of Integrative Natural Science
• /
• v.11 no.4
• /
• pp.192-196
• /
• 2018

The mobile agent paradigm can be exploited in a variety of ways, ranging from low-level system administration tasks to middle ware to user-level applications. Mobile agents can be useful in building middle-ware services such as active mail systems, distributed collaboration systems, etc. An active mail message is a program that interacts with its recipient using a multimedia interface, and adapts the interaction session based on the recipient's responses. The mobile agent paradigm is well suitable to this type of application, since it can carry a sender-defined session protocol along with the multimedia message. Mobile agent communication is possible via method invocation on virtual references. Agents can make synchronous, one-way, or future-reply type invocations. Multicasting is possible, since agents can be aggregated hierarchically into groups. A simple check-pointing facility has also been implemented. Another proposed solution is to use multi agent computer systems to access, filter, evaluate, and integrate this information. We will present the overall architectural framework, our agent design commitments, and agent architecture to enable the above characteristics. Besides, the each information needed a mobile agent system such as text, graphic, image, audio and video etc, constructed a great capacity multimedia database system. However, they have problems in establishing connections over multiple subnetworks, such as no end-to-end connections, transmission delay due to ATM address resolution, no QoS protocols. We propose a new connection management scheme in the thesis to improve the connection management involved of mobile agent systems.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.8 no.4
• /
• pp.1344-1367
• /
• 2014

Mobile traffic is increasing a masse because of the propagation of the Internet and the development of wireless mobile technology. Accordingly, the Network Local Mobility Management (NETLMM) working group [1] of the Internet Engineering Task Force (IETF) has standardized Proxy Mobile IPv6 (PMIPv6) [2] as a protocol for accomplishing the transmissibility of mobile terminals. PMIPv6 is a network-led IP-based mobility management protocol, which can control terminal mobility without depending on the type of access system or the capability of the terminal. By combining PMIPv6 and the mobility of Session Initiation Protocol (SIP), we can establish terminal mobility and session mobility through a more effective route. The mobility function can be improved and the overlap of function reduced as compared to that in the case of independent operation. PMIPv6 is appropriate for a non-real-time service using TCP, and SIP is appropriate for a real-time service using RTP/UDP. Thus, in the case of a terminal using both services, an effective mobility management is possible only by using PMIPv6 together with SIP. In order to manage mobility in this manner, researches on PMIPv6-SIP are in progress. In line with this trend, this paper suggests a new PMIPv6-SIP architecture where when a mobile terminal conducts a handover, a network-led handover while maintaining the session without the addition of a special function or middleware is possible along with effective performance evaluation through mathematical modeling by comparing the delay and the packet loss that occur during the handover to the Pure-SIP.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.19 no.7
• /
• pp.1568-1576
• /
• 2015

This paper proposes a prediction reputation system for the anti-SPIT solution in real-time VoIP environment. Increased accuracy of the determination as to whether spam or not by deriving a threshold based on SPIT presence in the existing paper. The existing schemes need to get the user's feedback and/or have experienced the time delay and overload as session initiates due to real-time operation. To solve these problems, the proposed scheme predicts the reputation through the statistical analysis based on the period of session initiation of each caller and the call duration of each receiver. As per the second mentioned problem, this scheme performs the prediction before session initiation, therefore, it's proper for real-time VoIP environment.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.27 no.5C
• /
• pp.445-453
• /
• 2002

We propose a cell-multiplexing scheme for the real-time communication service in ATM network and a new service discipline guarantee end-to-end delay based on pseudo-isochronous cell switching. The proposed scheme consists of two level frame hierarchy, upper and lower frame, which is used to assign the bandwidth and to guarantee the requested delay bound, respectively. Since the proposed algorithm employs two level frame hierarchy, it can overcome the coupling problem which is inherent to the framing strategy. The proposed scheme consists of two components, traffic controller and scheduller, as the imput traffic description model and regulates the input traffic specification. The function of the traffic controller is to shape real-time traffic to have the same input pattern at every switch along the path. The end-to-end delay is bounded by the scheduller which can limit the delay variation without using per-session jitter controllers, and therefore it can decrease the required buffer size. The proposed algorithm can support the QoS's of non-real time traffic as well as those of real time traffic