Browsing by Author "Chen, Whai-En"

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A performance study for IPv4-IPv6 translation in IP multimedia core network subsystem
(2010-08-01)
Chen, Whai-En
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Lin, Pin-Jen
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資訊工程學系
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Department of Computer Science
A Next Generation Network (NGN) is an advanced, packet-based network that exploits broadband and QoS-enabled transport technologies for enabling multimedia services. In NGNs, the principles and requirements of IP Multimedia Sub-system (IMS), which are used to deliver the desired benefits, should be carefully examined and studied. Internet Protocol version 6 (IPv6) is adopted by Third-generation Partnership Project (3GPP) to solve address storage problem and provide new features (e.g. plug-and-play and mobility) for IMS. However, in the early stage of IPv6 deployment, the existing Voice-over-IP (VoIP) networks support Internet Protocol version 4 (IPv4) only. For IPv4 IPv6 interworking between IMS and the existing VoIP networks, the IMS-Application Level Gateway (IMS-ALG) and the Transition Gateway (TrGW) are proposed to translate Session Initiation Protocol (SIP) and Real-time Transport Protocol (RTP) packets, respectively. In this paper, we focus on the IPv4 IPv6 translation for RTP packets, which is the bottleneck of VolP performance. Specifically, we developed a TrGW called National Information and Communications Initiative (NICI)-TrGW. In NICE-TrGW, we perform IPv4 IPv6 translation at the Linux kernel and adjust the header room of kernel-level packet buffer for each packet to reduce memory-copy operations. We evaluate the performance of NICI-TrGW and the existing solutions by using the SmartBits. Our study indicates that NICI-TrGW outperforms the existing solutions in terms of three different output measures including packet loss rate, maximum throughput, and average latency. Copyright (C) 2009 John Wiley & Sons, Ltd.
A Weakly Consistent Scheme for IMS Presence Service
(2009-07-01)
Chen, Whai-En
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Lin, Yi-Bing
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Liou, Ren-Huang
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資訊工程學系
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Department of Computer Science
IP Multimedia Core Network Subsystem (IMS) provides presence service for Universal Mobile Telecommunications System (UNITS). In IMS, the presence server is responsible for notifying an authorized watcher of the updated presence information. If the updates occur more frequently than the accesses of the watcher, the presence server will generate many notifications. This paper uses a weakly consistent scheme (called delayed update) to reduce the notification traffic. In this scheme, a delayed timer is defined to control the notification rate. We propose an analytic model and simulation experiments to investigate the performance of delayed update. The study indicates that delayed update can effectively reduce the notification traffic without significantly degrading the valid access probability.
An effective IPv4-IPv6 translation mechanism for SIP applications in next generation networks
(2010-08-01)
Chen, Whai-En
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Huang, Ya-Lin
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Lin, Yi-Bing
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資訊工程學系
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Department of Computer Science
In a next generation network, the IPv6-enabled IP multimedia subsystem (IMS) network may connect to an IPv4 network. When an IPv4/IPv6 dual-stack user equipment (UE) initiates a call by sending an IPv6 SIP INVITE message to an IPv4-only user agent (UA), the call cannot be established correctly. To resolve this problem, the IMS-application layer gateway solution, the redirect solution, and the interactive connectivity establishment solution have been proposed. In this paper, we propose an effective solution where only the IPv6 INVITE message is translated into an IPv4 INVITE message. Upon receipt of the IPv4 200 OK message replied from the IPv4-only UA, the dual-stack UE learns that the correspondent UA supports IPv4-only and utilizes IPv4 instead of IPv6 to send the subsequent SIP messages and real-time transport protocol (RTP) packets. The proposed solution is compared with the existing solutions in terms of network node modification, call setup complexity, and RTP transmission latency. Our study indicates that the proposed solution outperforms the other three solutions in the call setup and the RTP transmission. Copyright (C) 2009 John Wiley & Sons, Ltd.
Modeling VolP call holding times for telecommunications
(2007-11-01)
Chen, Whai-En
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Hung, Hui-Nien
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Lin, Yi-Bing
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交大名義發表
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National Chiao Tung University
Voice over IP is one of the most popular applications in broadband access networks. It is anticipated that the characteristics of call holding times (CHTs) for VoIP calls will be quite different from traditional phone calls. This article analyzes the CHTs for mobile VoIP calls based on measured data collected from commercial operation. Previous approaches directly used the Kolmogorov-Smirnov (K-S) test to derive the CHT distributions, which may cause inaccuracy. In this article we propose a new approach to derive the CHT distributions for mobile VoIP calls and other call types. Specifically, our approach uses hazard rate to select an appropriate distribution, and then utilizes the K-S test to validate our selection. We show that. the mobile VoIP CHT distribution can be accurately approximated by a mix of two log-normal distributions. Based on the derived distributions, we compare the mobile VoIP CHTs with those for non-VoIP calls and fixed-network VoIP calls. Our study indicates that the characteristics for mobile VoIP calls are quite different from those of the non-VoIP mobile phone calls and are more close to those of fixed-network phone calls.
NAT traversing solutions for SIP applications
(2008)
Chen, Whai-En
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Huang, Ya-Lin
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Chao, Han-Chieh
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資訊工程學系
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Department of Computer Science
Session Initiation Protocol (SIP) has been proposed for multimedia services and wide-area connectivity in smart home environments (SHEs). An important issue for SIP deployment in SHEs is network address translator (NAT) traversing. SIP and Real-time Transport Protocol (RTP) packets are delivered between an SHE (i.e., private IP network) and Internet (i.e., a public IP network) through an NAT function of a home gateway, and the NAT translates the IP/transport layer address and port number but leaves the application layer content unchanged. This results in inconsistency between the IP addresses/port numbers in the IP/transport layers and those in the SIP layer. To resolve this issue, we describe six solutions including static route, UPnP, STUN, ICE, ALG, and SBC. Then we compare these solutions in terms of smart home appliance (SHA) modification, scope of NATs supported, multilayer NAT traversal, ease of configuration, security issue, and time complexities. Copyright (c) 2008 Whai-En Chen et al.
PERFORMANCE MEASUREMENTS OF TD-LTE, WIMAX AND 3G SYSTEMS
(2013-06-01)
Lin, Yi-Bing
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Lin, Pin-Jen
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Sung, Yingrong Coral
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Chen, Yuan-Kai
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Chen, Whai-En
Long Term Evolution (LTE) and Worldwide Interoperability for Microwave Access (WiMAX) are two major technologies toward next-generation mobile broadband standards, which are both expected to provide higher throughput and lower transmission latency for mobile users. This article measures and compares the latency and the throughput of Time Division - Long Term Evolution (TD-LTE), WiMAX, and third-generation (3G) systems based on several technical trials. Our quantitative measures and comparisons provide guidelines for the operators to deploy their future networks.
Real-Time VoIP Quality Measurement for Mobile Devices
(2011-12-01)
Chen, Whai-En
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Lin, Pin-Jen
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Lin, Yi-Bing
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資訊工程學系
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Department of Computer Science
The quality of voice over Internet protocol (VoIP) is typically measured by mean opinion score (MOS) that significantly varies in wireless environments. Several approaches have been proposed to evaluate the MOS values of voice paths. However, these solutions require extra servers/gateways to conduct heavy computation for MOS measurement of VoIP calls. We propose two MOS measurement procedures for mobile devices. In our approach, a simple program is installed in each of the mobile devices (through an app-store like mechanism). In the perceptual evaluation of speech quality (PESQ) MOS measurement procedure, a lightweight real-time table lookup solution significantly reduces the computation time of PESQ MOS measurement from 315.4 s to 3 s. In the E-model MOS measurement procedure, the E-model MOS value can be accurately computed in 5.35 s.
SIPv6 analyzer: an analysis tool for 3GPP IMS services
(2008-02-01)
Chen, Whai-En
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Sung, Yueh-Hsin
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Lin, Yi-Bing
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資訊工程學系
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Department of Computer Science
The 3rd Generation Partnership Project (3GPP) defines IP Multimedia Core Network Subsystem (IMS) to support IP-based multimedia services. IMS utilizes protocols such as IP version 6 (IPv6), Session Initiation Protocol (SIP), and Real-time Transport Protocol (RTP) to deliver the multimedia content. This paper proposes an analysis tool referred to as SIPv6 Analyzer to investigate the IMS-related protocols during IMS service deployment. The SIPv6 Analyzer not only dissects the protocol headers but also provides user-friendly functions such as message flow generation and audio/video replay for IPv6, SIP, and RTP. We describe the design of the SIPv6 Analyzer and use examples to illustrate how it works. Copyright (C) 2006 John Wiley & Sons, Ltd.