• Title/Summary/Keyword: Mobile VoIP

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A New Mobile-IPv6 based Buffering Scheme in the All-IP Network

  • Park, Byoung-Seob;Lim, Cheol-Su
    • Proceedings of the IEEK Conference
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    • 2002.07b
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    • pp.1094-1097
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    • 2002
  • Realtime applications like VoIP(Voice over IP) in All-IP networks need smooth handoffs in order to minimize or eliminate packet loss as a Mobile Host(MH) transitions between network links. In this paper, we design a new dynamic buffering(DB) mechanism for IPv6 by which an MH can request that the router on its current subnet buffers packets on its behalf while the MH completes registration procedures with the router of a new subnet. Performance results show that our proposed buffering scheme with a dynamic buffer space allocation is quite appropriate for mobile Internet, or the All-IP environment in terms of the datagram loss rate.

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A Study on Secure and Efficient Key Distribution for Group Communication (안전하고 효율적인 그룹 키 분배에 관한 연구)

  • Kim, Jung-Yoon;Choi, Hyoung-Kee
    • Proceedings of the Korean Society of Computer Information Conference
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    • 2009.01a
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    • pp.373-376
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    • 2009
  • 최근 네트워크 기술의 발전에 의해 VoIP, IP-TV 등 다양한 서비스들이 등장하였다. 이러한 실시간 서비스들은 품질을 보장하면서 통신 내용을 안전하게 보호할 수 있는 보안 메커니즘이 필수적이다. 우리는 VoIP를 이용한 다자간 통화 및 화상 회의, 그리고 IP-TV와 같은 그룹 기반의 서비스들을 안전하고 효율적으로 보호하기 위한 그룹 카 분배 프로토콜을 제안한다. 제안하는 프로토콜은 빠르고 효율적인 연산만으로 구성되었으며, 그룹 내부 및 외부의 공격으로부터 그룹 키를 안전하게 보호한다. 성능평가 및 분석 결과는 제안하는 프로토콜이 최근에 연구된 다른 프로토콜들에 비해 안전하고 효율적임을 증명하였다.

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Enhanced Timing Recovery Using Active Jitter Estimation for Voice-Over IP Networks

  • Kim, Hyoung-Gook
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.6 no.4
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    • pp.1006-1025
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    • 2012
  • Improving the quality of service in IP networks is a major challenge for real-time voice communications. In particular, packet arrival-delay variation, so-called "jitter," is one of the main factors that degrade the quality of voice in mobile devices with the voice-over Internet protocol (VoIP). To resolve this issue, a receiver-based enhanced timing recovery algorithm combined with active jitter estimation is proposed. The proposed algorithm copes with the effect of transmission jitter by expanding or compressing each packet according to the predicted network delay and variations. Additionally, the active network jitter estimation incorporates rapid detection of delay spikes and reacts to changes in network conditions. Extensive simulations have shown that the proposed algorithm delivers high voice quality by pursuing an optimal trade-off between average buffering delay and packet loss rate.

The Header Compression Scheme for Real-Time Multimedia Service Data in All IP Network (All IP 네트워크에서 실시간 멀티미디어 서비스 데이터를 위한 헤더 압축 기술)

  • Choi, Sang-Ho;Ho, Kwang-Chun;Kim, Yung-Kwon
    • 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.

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M_DQDMR Algorithm for Improved QoS of Mobile VoIP Services (Mobile 환경에서 VoIP 서비스의 QoS 향상을 위한 M_DQDMR 알고리즘)

  • 서세영;최승권;신승수;조용환
    • Proceedings of the Korea Contents Association Conference
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    • 2003.05a
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    • pp.383-389
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    • 2003
  • In this paper, we propose a solution, called M_DQDMR, for generating delay-constrained boundwidth-appropriated multicast routing trees to reduce the delay and conserved boundwidth resources of nodes in mobile computing environment. At the current router installation, we according the routing table and the information of link which neighboring rude to guarantee QoS(Quality of Service). When we construct multicast tree, M_DQDMR algorithm dynamically adjusts its appropriate tree construction policy based on how far the destination node from the delay bound and boundwidth our QoS requirement. Through simulations and comparing to another multicast algorithm, we reach a conclusion is that M_DQDMR can simply and dynamically adjusts the construction of multicast tree in hight-speed and conserve boundwidth resources.

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A Semi-Soft Handoff Mechanism with Zero Frame Loss in Wireless LAM Networks (무선 LAN 환경에서 프레임 손실 없는 Semi-Soft 핸드오프 방안)

  • 김병호;민상원
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.28 no.12B
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    • pp.1135-1144
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    • 2003
  • In this paper, we proposed a semi-soft handoff mechanism to provide link mobility in IEEE 802.11 wireless LAN environment. Buffers and routing tables in APs and portals are provided in order to reroute frames, which have not been received during handoff time and have been buffered in an old AP, to a new AP after handoff is performed. For the re -routing operation, the MAC routing table should be updated by exchanging information of a mobile terminal between neighbor APs. With our proposed scheme. a wireless LAN node can perform semi soft handoff while changing its attached AP and provide mobile IP and/or real time service like voice over IP. Also, we have done simulation for evaluation of the performance of the proposed scheme. We show that our semi soft handoff mechanism can be applied for real-time service with no frame loss in mobile environment.

A Design of Video Phone System based on SIP for HomeServer (SIP 기반의 홈서버용 영상전화 시스템 설계)

  • Ahn, Sung-Ho;Lee, Kyung-Hee;Lee, Eun-Ryoung;Kim, Ji-Yong;Kim, Doo-Hyun
    • Proceedings of the Korea Information Processing Society Conference
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    • 2002.11a
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    • pp.53-56
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    • 2002
  • VoIP(Voice over IP) 기술은 다양한 인터넷 응용 서비스 보급의 대중화에 기여한 주요 통신기술의 하나이다. 이에 따라 인터넷 이용자가 급격히 증가하고 인터넷 전화의 수요가 증가하게 되었다. 특히, 음성 뿐만 아니라 영상에 대한 기술이 접목되어 V2oIP(Voice and Video over IP) 라 일컬어 지는 기술이 보급되면서, 인터넷 영상전화에 대한 대중화가 이루어 지고 있다. 한편, 다양한 인터넷 응용 서비스 보급의 대중화에 따른 가정 내에서의 네트워크 환경 또한 부각되고 있어 정보가전분야에 큰 변화가 일고 있다. 이에 홈서버 중심의 홈네트워크환경이 구축된다. 따라서, 기존의 pc 를 단말기로 한 인터넷 환경 및 제반 응용 서비스가 그대로 홈서버 중심의 홈네트워크환경으로 옮겨져야 할 필요가 있다. 본 논문에서의 전체 시스템은 임베디드 리눅스 기반인 Qplus 운영체제를 기반으로 하는 홈서버상에 HoCoS(Home Collaboration Service) 시스템이 탑재되며, 이 시스템은 크게 영상전화 시스템과 공동작업 시스템으로 구성된다. 본 논문에서는 상기 시스템 중 SIP(Session Initiation Protocol)기반의 홈서버용 영상전화 시스템에 대한 설계 및 구현에 대해 기술하고자 한다.

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Design and Implementation of effective ECC Encryption Algorithm for Voice Data (음성 데이터 보안을 위한 효율적인 ECC 암호 알고리즘 설계 및 구현)

  • Kim, Hyun-Soo;Park, Seok-Cheon
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.15 no.11
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    • pp.2374-2380
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    • 2011
  • Many people is preferred to mVoIP which offers call telephone-quality and convenient UI as well as free of charge. On the other hand, security of mVoIP is becoming an issue as it using Internet network may have danger about wiretapping. Although traditionally encryption algorithm of symmetric key for security of voice data has been used, ECC algorithm of public key type has been preferring for encryption because it is stronger in part the strength of encryption than others. However, the existing way is restricted by lots of operations in poor mobile environment. Thus this paper proposes the efficiency of resource consumption way by reducing cryptographic operations.

Proposal of Traffic Management Strategy between Hierarchical Mobile-WiMAX/WLAN Networks (계층적 Mobile-WiMAX/WLAN 네트워크에서의 트래픽 관리 전략에 관한 연구연구)

  • Moon, Tae-Wook;Kim, Moon;Cho, Sung-Jun
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.13 no.1
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    • pp.153-160
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    • 2009
  • A hierarchical network between Mobile-WiMAX and WLAN systems is to make it possible to utilize wireless internet services including time sensitive applications such as VoIP, VOD, visual telephony etc. During the process of vertical handoff decision from Mobile-WiMAX to WLAN hotspot, vertical handoff delay causes user dissatisfaction because it doesn't provide the seamless wireless internet service. We make use of type of service (ToS) parameters in IEEE 802.16e specification as the criterion parameter of vertical handoff decision process in hierarchical Mobile-WiMAX/WLAN networks. In this paper, we propose the process of vertical handoff decision for seamless wireless internet service which is sensitive to time delay. If type of service is time sensitive application, the decision of vertical handoff is withdrawn until the service is terminated. In focus on user satisfaction, if the proposed traffic management strategy in hierarchical Mobile-WiMAX/WLAN networks is used, user will utilize seamless wireless internet services including time sensitive applications.