• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.11-14
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• 2005

In this paper, we propose a method for transmitting SMS(Short Message Service) for VoIP(Voice over IP) service supporting multi-protocol. The multi-protocol VoIP under consideration are generally composed of H.323, SIP and MGCP and Most ITSPs(Internet Telephony Service Provider) provide VoIP service with H.323 and SIP now. SMS is killer application in mobile telecom service and many people of various field use that service. For example, user can send many SMS messages and substitute e-mail. Also SMS can be provided with various internet application. Ahn, legacy phone of KT, can use SMS. Therefore VoIP phone also can be required with the same requirement. With the multi-protocol VoIP we will propose several methods sending efficiently SMS. To show the validity of the proposed method some examples are given in which the results can be expected by intuitive observation.

• Journal of Advanced Navigation Technology
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• v.17 no.4
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• pp.421-428
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• 2013

Voice over IP refers to technology that enables routing of voice conversations over the Internet or a TCP/IP network. VoIP communication costs cheaper than traditional analog phone. Phone calls can be made to anywhere / anyone: Both to VoIP numbers as well as people with normal phone numbers. VoIP protocol equipment available today follows the SIP standard. Older VoIP equipment though would follow H 323, MGCP, Megaco/H.248. A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the TCP/IP network. A SIP server is also referred to a Asterisk IP-PBX. A VoIP telephone, also known as a SIP phone or a softphone, allows the user to make phone calls to any softphone, mobile or PC by using App store. A VoIP telephone can be a simple software-based softphone. However, the SIP Server and the program is vulnerable to VoIP attacks. In this paper, eavesdropping attacks tested by using the Asterisk SIP server. Eavesdropping attacks and TLS security methods apply to VoIP system. TLS can be applied to determine whether the eavesdropping available for VoIP Environments.

• Journal of the Korea Convergence Society
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• v.3 no.3
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• pp.7-12
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• 2012

mVoIP (mobile Voice over Internet Protocol) service is a technology to transmit voice data through an IP network using mobile device. mVoIP provides various supplementary services with low communication cost. It can maximize the availability and efficiency by using IP-based network resources. In addition, the users can use voice call service at any time and in any place, as long as they can access the Internet on mobile device easily. However, SIP on mobile device is exposed to IP-based attacks and threats. Observed cyber threats to SIP services include wiretapping, denial of service, and service misuse, VoIP spam which are also applicable to existing IP-based networks. These attacks are also applicable to SIP and continuously cause problems. In this study, we analysis the threat and vulnerability on mVoIP service and propose several possible attack scenarios on existing mobile VoIP devices. Based on a proposed analysis and vulnerability test mechanism, we can construct more enhanced SIP security mechanism and stable mobile VoIP service framework after eliminating its vulnerability on mobile telephony system.

• Journal of Institute of Control, Robotics and Systems
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• v.22 no.8
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• pp.671-677
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• 2016

A Controller Area Network (CAN) is a serial communication protocol that is highly reliable and efficient in many aspects, such as wiring cost and space, system flexibility, and network maintenance. Therefore, it is chosen for the communication protocol between a single chip controller based on Field Programmable Gate Array (FPGA) and peripheral devices. In this paper, the design and implementation of CAN IP, which is written in VHSIC Hardware Description Language (VHDL), is presented. The implemented CAN IP is based on the CAN 2.0A specification. The CAN IP consists of three processes: clock generator, bit timing, and bit streaming. The clock generator process generates a time quantum clock. The bit timing process does synchronization, receives bits from the Rx port, and transmits bits to the Tx port. The bit streaming process generates a bit stream, which is made from a message received from a micro controller subsystem, receives a bit stream from the bit timing process, and handles errors depending on the state of the CAN node and CAN message fields. The implemented CAN IP is synthesized and downloaded into SmartFusion FPGA. Simulations using ModelSim and chip test results show that the implemented CAN IP conforms to the CAN 2.0A specification.

• Journal of Internet Computing and Services
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• v.6 no.1
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• pp.13-25
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• 2005

Distributed Denial-of-Service(DDoS) attack prevent users from accessing services on the target network by spoofing its origin source address with a large volume of traffic. The objective of IP Traceback is to determine the real attack sources, as well as the full path taken by the attack packets. Existing IP Traceback methods can be categorized as proactive or reactive tracing. Existing PPM based tracing scheme(such as router node appending, sampling and edge sampling) insert traceback information in IP packet header for IP Traceback. But, these schemes did not provide enhanced performance in DDoS attack. In this paper, we propose a 'TTL based advanced Packet Marking' mechanism for IP Traceback. Proposed mechanism can detect and control DDoS traffic on router and can generate marked packet for reconstructing origin DDoS attack source, by which we can diminish network overload and enhance traceback performance.

• Journal of Institute of Control, Robotics and Systems
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• v.8 no.5
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• pp.434-437
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• 2002

A small size and light-weight DSP board is newly designed for a real time multi-distributed control system that overcomes constraints on time and space. There are a variety of protocols for a real-time distributed control system. In this research, we selected CAN for the multi distributed control, which was developed by the BOSCH in the early 1980's. If CAN and Internet are connected together, the system attains the characteristics of a distributed control system and a remote control system simultaneously. To build such a system. The TCP/IP-CAN Gateway which converts a CAN protocol to TCP/IP protocol and vice verse, was designed. Moreover, the system is required to be small and light-weighted for the high mobility and cost effectiveness. The equipment in remote place has a TCP/IP-CAN Gateway on itself to be able to communicate with another systems. The received commands in the remote site are converted from TCP/IP protocol to CAN protocol by the TCP/IP-CAN Gateway in real time. A simulation system consists of a TCP/IP-CAN Gateway in remote place and a command PC to be connected to Ethernet.

• 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.

• Journal of the Korea Society of Computer and Information
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• v.17 no.3
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• pp.113-120
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• 2012

VoIP provides voice data service using existing IP networks and has received much attention recently. VoIP service has a variety of security vulnerabilities. Types of main attacks on VoIP service are tapping/interception, DoS attacks, spam, misuse of service attacks and the like. Of these, confidential information leak because of tapping/interception has been considered as a critical problem. Encryption techniques, such as SRTP and ZRTP, are mostly used to prevent tap and intercept on VoIP media information. In general, VoIP service has two service scenarios. First, VoIP service operates within a single private network. Second, VoIP service operates between different private networks. Both SRTP and ZRTP for VoIP media information within a single private network can perform encryption. But they can not perform encryption between different private networks. In order to solve this problem, in this paper, we modify SRTP protocol. And then, we propose an encryption method that can perform encryption of VoIP media information between the different private networks.

• Journal of Digital Contents Society
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• v.3 no.2
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• pp.187-196
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• 2002

When mobile node moves into the different IP address area, the mobile node can be set the proper environment parameters automatically by DHCP. If we extend the DHCP option field to support the roles of home agent and foreign agent in mobile IP protocol, mobile node can perform the foreign address registration process of mobile IP protocol when DHCP IP environment is achieved automatically, too. DHCP is supplied in the PC with a series of Windows NT. This paper proposes and implements the functional blocks of extended DHCP nodes and this enables the DHCP node have a role of mobile IP agents without another protocol functional blocks.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.6B
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• pp.575-584
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• 2009

In this paper, we design and implement a VoIP traffic generator for simulating call processing in IP contact center systems. Creating a VoIP call based on H.323 and SIP and generating RTP traffic which uses G.711 codec, the generator lets many users simulate situations on which they call each other. With this tool, which is named VoIPTG, users can combine H.323 or SIP session control protocol, the number of users, time variation, and voice codecs and then direct various situations for simulation. This traffic generator can be used for testing functions of an IP contact center and especially it is necessary for testing the quality of IP based call recording systems.