• 한국정보통신설비학회:학술대회논문집
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• 2007.08a
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• pp.331-335
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• 2007

방송과 초고속인터넷 서비스를 동시에 제공할 수 있는 HFC(Hybrid Fiber Coaxial) 망은 상/하향이 비대칭 구조이며, 하향속도에 비해 상향속도가 1/10 수준이어서 상향 트래픽이 과다하게 생성될 경우 인터넷속도 지연이 발생한다. 지연에 민감한 VoIP 서비스의 품질보장 방안으로는, DOCSIS(Data Over Cable System Interface Specification) 1.1 기반의 상향 스케쥴링 기능을 이 용한 VoCM(Voice Over Cable Modem)이 있다. 그러나 별도의 VoCM을 사용해야 하며 아날로그 전화기를 사용해 IP 기반의 VoIP 단말을 사용할 수 없다는 단점이 있다. 일반 CM(Cable Modem)에 DOCSIS 1.1 Config File을 이용하여 VoIP 품질을 보장할 경우 별도의 트래픽 대역을 항상 점유해야 하는 단점이 있다. 이에, 본 논문에서는 효율적 대역폭 이용과 단말장비에 종속적이지 않은 방안을 제안하고 일반 CM을 통한 유무선 환경하에서 Dynamic QoS(Quality Of Service)를 제공할 수 있는 PCMM(Packet Cable MultiMedia) 적용 방안 및 시험결과에 대해 고찰하고자 한다.

• Journal of Information Technology Applications and Management
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• v.18 no.2
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• pp.23-38
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• 2011

The advance of All-IP digital convergence is now triggering fundamental changes in the media industry. This article analyzes the evolution of the media industry based on a value chain model, with a special emphasis placed on the impact of digital technology on the structure of media production, distribution, and consumption. As the Internet has evolved into an access channel for all forms of media, the boundaries of the media industry remain unclear and open thus enabling anyone to become a creator of media. Furthermore, the scope of media continues to become more dynamic as competition grows between older and more innovative forms of media. In this light, adapting to innovative technologies and more effectively meeting the needs of customers represent key factors for the continued evolution of the media.

• Journal of Korea Multimedia Society
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• v.10 no.3
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• pp.335-347
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• 2007

VoIP (Voice over IP) is a technology to transport video and voice traffic over IP networks such as Internet. Today, the VoIP technology is viewed as the right choice for providing voice, video, and data communication among various terminals over the next generation network. This paper discusses a multipoint VoIP implementation with end-point mixing model which can support multipoint conference without a conference bridge. The multipoint VoIP is implemented with SIP (Session Initiation Protocol), and supports STUN (Simple Traversal of UDP Through NATs) since it works in an asymmetric NAT (Network Address Translator) environment. The characteristics of this paper are as follows. It is possible that all terminals in the hierarchical conference don't receive the duplicated media information because we use the end-point mixing model with the new media processing module. And, the paper solves the problem that the hierarchical conference session should be separated into several sessions when a mixing terminal terminates the hierarchical conference session.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.49-54
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• 2009

Synchronization between media is an important aspect in the design of multimedia streaming system. This paper proposes a precise media synchronization mechanism for digital video and audio transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, RTP/RTCP protocol suite is usually employed. To provide a precise mechanism for media synchronization between video and audio, we suggest an efficient media synchronization algorithm based on NPT (Normal Play Time) which can be derivable from the timestamp information in the header part of RTP packet generated for the transport of video and audio streams. With the proposed method, we do not need to send and process any RTCP SR (sender report) packet which is required for conventional media synchronization scheme, and accordingly could reduce the number of required UDP ports and the amount of control traffic injected into the network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.11B
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• pp.991-1000
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• 2004

This paper presents an application layer multicast tree constructing algorithm to minimize the average time delay from the sender to end-systems for the effective real-time media delivery. Simultaneously, the proposed algorithm takes into account the computing power and the network condition of each end-system as a control variable and thus avoids the undesirable case that loads are concentrated to only several end-systems. The multicast tree is constructed by clustering technique and modified Dijkstra's algorithm in two steps, i.e. tree among proxy-senders and tree in each cluster. By the experimental results, we show that the proposed algorithm can provide an effective solution.

• International Journal of Internet, Broadcasting and Communication
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• v.13 no.1
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• pp.168-179
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• 2021

With the introduction of China's new policies, the Chinese animation industry has gradually formed a sustainable industrial structure chain, and the output value of the animation market is also in a state of continuous growth. Since 2013, domestic animation has been developing from "lower age" to "ageing" and "adults". At the same time, with the popularization of China's pan-entertainment industry model, the multi-domain symbiosis of the Internet and mobile Internet has been realized, creating a fan economy of star IP (Intellectual Property), and promoting the linkage of various industries under the same IP. This paper mainly analyzes the development of the animation IP market in China's pan-entertainment mode in recent years, and analyzes the cross-media operation mode of the animation industry. At the same time, it studies the application of self-media in animation.

• The Journal of the Korea Contents Association
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• v.4 no.2
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• pp.43-49
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• 2004

SIP Caller Preference is an useful function that allows a caller to express preferences about request handling in servers. It can also feat appropriate call processing according to the callee capabilities. However, only the category of the media is considered as a criteria for target selection in the caller preference. In this case, if the callee's media information such as codec is different from the caller, an additional re­negotiation occurs for SIP call setup. Therefore, in this paper, we have suggested an extended caller preference to solve this problem. In our SIP based VoIP system, a network sewer uses detailed media informations for media stream in the session to select the target for SIP call setup. The sewer gives higher priority to the candidate which do not require re-negotiation for call setup, so that an effective call setup can be achieved in our system.

• Journal of the Korea Society of Computer and Information
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• v.18 no.4
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• pp.87-96
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• 2013

In this paper, we suggest redesigned RTP protocol that is able to perform encryption of VoIP media information for single private network and between the different private networks. And we conduct a test for performance analysis. Such as SRTP or ZRTP methods have been used for VoIP media encryption. But, the existing encryption techniques have problem that can not perform end-to-end encryption between different private networks. In order to solve this problem, in this paper, we redesign RTP protocol. Redesigned RTP includes all information for encryption of VoIP media. Therefore the encryption is not affected by modification of SIP and SDP information that occurred in gateway. Also, redesigned RTP includes code for whether or not to apply encryption. By using the code, modification of RTP header from gateway prevents. As a result, redesigned RTP maintain the integrity and the RTP is able to perform encryption between the different private networks. Also, we conduct a test for performance analysis of SRTP, ZRTP and redesigned RTP.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.11a
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• pp.839-843
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• 2002

In this paper, we present a next generation VoIP network architecture. Specifically, we present key components such as SoftSwitch and media gateway, and important protocols for VoIP services. We also present basic components and mechanisms for VoIP QoS. The architecture presented in this paper is able to support open interfaces and multimedia traffic, and therefore various multi-services can be supported efficiently using the architecture.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.18 no.3
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• pp.109-120
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• 2008

VoIP utilizes the Internet for the services, and therefore it is vulnerable to intrusions and attacks. Because provided services deal with information related to privacy of users, it requires high level security including authentication and the confidentiality/integrity of signaling messages and media streams. However, when such a protocol is implemented in a VoIP phone, the implementation can have limitations due to the limited resources. The present study purposed to implement VoIP security protocols and to evaluate their performance in terms of connection quality and voice quality by applying them to SIP proxy and UA (User Agent). In the result of performance evaluation, the application of the security protocols did not lower voice quality, but connection quality was high in the DTLS based security protocol. As the protocol was applicable to signaling and media paths based on DTLS, we found that it can be a solution for the limited resources of VoIP phone.