• The Transactions of the Korea Information Processing Society
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• v.4 no.12
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• pp.3088-3096
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• 1997

In this paper, we present WWW system over native ATM services. The use of native ATM services through ATM API can provide better performance and functionality than that of IP over ATM, LAN Emulation or Multiprotocol over ATM. Our WWW browser and server provide advanced WWW services based on enhanced performance and guaranteed QoS support by using native ATM service benefits. This paper describes and compares advantages and disadvantages of Native ATM Services and ATM Internet Services, and addresses ATM API standardization and development trend that are made by the ATNI Forum for the support of native ATM services, and then describes the architecture and operation of our WWW browser and server using ATM API. The system architecture is based on HTTP over ATM API capable of supporting guaranteed QoS over its connections. The system defines and uses new HTML attributes within hyperlinking HTML elements for the description of ATM QoS and traffic characteristics that are derived from UNI signaling 3.1 connection characteristics information elements. Our system uses WinSock 2 API as its ATM API.

• Journal of KIISE:Information Networking
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• v.30 no.5
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• pp.621-630
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• 2003

In this paper, we implemented SIP protocol that supports IPv6 and differentiated call processing scheme for NGN(Next Regeneration Network). In NGN, SIP processes call signaling among various application services. A softswitch and SIP server must give priority to sensitive services such as Fax, network management and home networking that require a fast call setup time. Also, the support of IPv6 is needed under consideration of All-IP. We proposed differentiated call processing scheme. The differentiated call processing scheme supports differentiated call processing as priority of service class on call processing in SW server We defined three service classes and use the Flow Label field of the IPv6 header for setting service class. Through the performance analysis, we proved that it improves throughput for call message with the high priority. The result of performance analysis demonstrates that differentiated call processing scheme gives better performance for the service requiring a fast session establishment in NGN.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.19 no.3
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• pp.73-82
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• 2009

This paper proposes an efficient and secure user authentication and key agreement scheme instead of the HTTP digest and TLS between the SIP UA and server. Although a number of security schemes for authentication and key exchange in SIP network are proposed, they still suffer from heavy computation overhead on the UA's side. The proposed scheme uses the HTIP Digest authentication and employs the Diffie-Hellman algorithm to protect user password against dictionary attacks. For a resource-constrained SIP UA, the proposed scheme delegates cryptographically computational operations like an exponentiation operation to the SIP server so that it is more efficient than the existing schemes in terms of energy consumption on the UA. Furthermore, it allows the proposed scheme to be easily applied to the deployed SIP networks since it does not require major modification to the signaling path associated with current SIP standard.

• Journal of Digital Contents Society
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• v.17 no.6
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• pp.581-592
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• 2016

WebRTC has quickly grown to be the world's advanced real-time communication in several platforms such as web and mobile. In spite of the advantage, the current technology in WebRTC does not handle a big-streaming efficiently between peers and a large amount request of users on the Signaling server. Therefore, in this paper, we put our work to handle the problem by delivering the flow of data with dynamical load balancing algorithms. We analyze the request source users and direct those streaming requests to a load balancing component. More specifically, the component determines an amount of the requested resource and available resource on the response server, then it delivers streaming data to the requesting user parallel or alternately. To show how the method works, we firstly demonstrate the load-balancing algorithm by using a network simulation tool OPNET, then, we seek to implement the method into an Ubuntu server. In addition, we compare the result of our work and the original implementation of WebRTC, it shows that the method performs efficiently and dynamically than the origin.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.9C
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• pp.842-852
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• 2002

The fast advance in the VoIP technologies gives a rich opportunity to create different kind of VoIP applications such as IP telephony services. The application level call signaling protocols such as ITU-T H.323 and IETF SIP provide the communication functions of end-to-end call setup and release. Currently, there is a lot of H.323 based VoIP products in the market, however SIP is considered as a suitable protocol for supporting applications in IP environments, so SIP based VoIP products and services begin to appear. In this paper, firstly we present the characteristics of some possible SIP based applications and describe the design and implementation of a VoIP example service named PC-to-PC Internet telephony service using the developed SIP network components. The PC-to-PC Internet telephony service and User Agent are developed in MS window 98/2000 using visual C/C++, and Proxy server and Registrar in Linux 7.0 using C, respectively.

• The Journal of Korean Association of Computer Education
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• v.12 no.4
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• pp.47-55
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• 2009

There are many researches considering mobile devices as resources in mobile grids. However, the mobile device has some limitations: wireless connection and battery capacity. So, the grid operations using mobile devices have lower reliability and efficiency than those in fixed grid environments. In this paper, we propose a job migration scheme using mobile devices to overcome these limitations. The proposed job migration scheme predicts failure condition during execution and takes checkpoints. Then, if the failure occurs on mobile device during execution, the executing job can be migrated to other mobile device by checkpoint information. To perform the proposed migration scheme, we establish a mobile device manager on a proxy server and a status manager on a mobile device. Connection, wireless signal strength and battery capacity of mobile devices are identified through two managers. The simulation results show improvement of efficiency and reliability during execution.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.11
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• pp.1481-1489
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• 2016

A art from the typical data services based on 1:1 data communications, various new services based on 1:N communications have recently appeared. These services are becoming to require advanced 1:N communication schemes which can transfer the same data to many receivers efficiently and in high-performance. Especially, a Push-To-Talk (PTT) service, which is an important service in public safety communication system, requires a service server to disseminate the same voice media data to multiple receivers in a group in real-time and low latency. In this paper, we propose an efficient scheme to disseminate the same data to multiple receivers in low latency. In addition, we provide an analysis which gives a guide the performance of the 1:N communications in practical wired/wireless system environments in the perspective of the PTT service index.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.6
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• pp.75-84
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• 2014

In order to solve the problems such as overload, single point of failure, non-optimized data path, and network scalability in conventional central mobility management protocols, distributed mobility management schemes have been continually studied in and around the IETF. In this paper, a network-based partially-distributed mobility management mechanism, pDMMv6, is suggested and the performance comparison with traditional protocols such as PMIPv6 and MIPv6 is made through simulation under the various user traffic environment. The simulation results include UDP packet delivery ratio, end-to-end packet delay, binding delay for registration signaling, CPU utilization in each node, and response delays in several server-client TCP applications such as web browsing, e-mail, telnet remote login, FTP file up/down-load, and database access.

• Journal of Digital Convergence
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• v.18 no.5
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• pp.249-256
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• 2020

The session initiation protocol (SIP) is a signaling protocol, which is used to controlling communication session creation, manage and finish over Internet protocol. Based on it, we can implement various services like voice based electronic commerce or instant messaging. Recently, Qiu et al. proposed an enhanced password authentication scheme for SIP. However, this paper withdraws that Qiu et al.'s scheme is weak against the off-line password guessing attack and has denial of service problem. Addition to this, we propose an improved password authentication scheme as a remedy scheme of Qiu et al.'s scheme. For this, the proposed scheme does not use server's verifier and is based on elliptic curve cryptography. Security validation is provided based on a formal validation tool ProVerif. Security analysis shows that the improved authentication scheme is strong against various attacks over SIP.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.22 no.3
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• pp.463-472
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• 2012

This paper propose a security method that performs mutual authentication between the SIP UA and the server, check for integrity of the signaling channel and protection of SDP information for VoIP using a One-Time Password. To solve the vulnerability of existing HTTP Digest authentication scheme in SIP, Various SIP Authentication schemes have been proposed. But, these schemes can't meet security requirements of SIP or require expensive cryptographic operations. Proposed method uses OTP that only uses hash function and is updated each authentication. So Proposed method do not require expensive cryptographic operations but performs user authentication efficiently and safely than existing methods. In addition, Proposed method verifies the integrity of the SIP messages and performs SDP encryption/decryption through OTP that used for user authentication. So Proposed method can reduce communication overhead when applying S/MIME or TLS.