Whai-En Chen

Design of multi-field IPv6 packet classifiers using ternary CAMs

Typically, high-end routers/switches classify a packet by looking for multiple fields of the IP/TCP headers and recognize which flow the packet belongs to. Several packet classification algorithms to accelerate packet processing and reduce the memory requirement have been proposed. But it is not easy to implement these algorithms in hardware to lookup these multiple fields in the same time. This paper intends to design a novel packet classification engine capable of simultaneously processing multi-field searching, especially for the IPv6 packets with relative longer addresses (128 bits). To classify the IPv6 packets in wire-speed, the CLM (CAM-Like Memory)-based hardware architecture is considered and five fields (source IPv6 address, destination IPv6 address, source port, destination port, and protocol) are designed as the searching key. Evaluation results indicate that compared with the typical market leading delivering search engines, the proposed hardware architecture provides a 30% speed-up performance. A compact method is also provided to compress the bit-width required to represent the multi-field of an IPv6 packet. This saves the memory space required for the IPv6 rule table for about 20%.

RSVP extensions for real-time services in hierarchical mobile IPv6

The Mobile IPv6 (MIPv6) provides many great features, such as sufficient addressing space, mobility, and security; MIPv6 is one of the most important protocols for next generation mobile Internet. Simultaneously, with the rapid improvement of wireless technologies, the real-time multi-media IP services such as video on demand, videoconference, interactive games, IP telephony and video IP phone will be delivered in the near future. Thus, to furnish accurate QoS for real-time services is one of the most important thing in the next generation mobile Internet. Although RSVP, which is a resource reservation protocol, processes signaling messages to establish QoS paths between senders and receivers, RSVP was originally designed for stationary networks and not aware of the mobility of MNs. Therefore, this paper proposes a novel RSVP extension to support real-time services in Hierarchical Mobile IPv6 (HMIPv6) environments. For intra-site mobility, the concept of QoS Agent (QA) is proposed to handle the RSVP QoS update messages and provide the advanced reservation models for real-time services. For inter-site mobility, IP multicast can help to invite inter-site QAs to make pre-reservation and minimize the service disruption caused by re-routing the data path during handover. Simulation results show that the proposed scheme over HMIPv6 is more suitable for real-time services than the famous RSVP tunnel-based solution.

Modeling VoIP Call Holding Times for Telecommunications

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.

WGSN: WLAN-based GPRS Support Node with Push Mechanism

This paper proposes wireless LAN (WLAN)-based GPRS Support Node (WGSN), as a solution for integrating Third Generation (3G) and WLAN services. We show that the 3G mechanisms can be re-used for WLAN user authentication and network access without introducing new procedures and without modifying the existing 3G network components. We describe the WGSN features and show how they are designed and implemented. To reduce the power consumption and computation complexity of a mobile station (MS), the WGSN applications may not be activated in the MS if they are not used. For an MS-terminated application, a push mechanism is implemented in WGSN, which automatically activates the application at the MS side. An analytic model is proposed to investigate the requirements on the WGSN transmission delay of the push operation. A WGSN prototype has been implemented in the Industrial Technology Research Institute and National Chiao-Tung University Joint Research Center.

A weakly consistent scheme for IMS presence service

IP multimedia core network subsystem (IMS) provides presence service for universal mobile telecommunications system (UMTS). 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 IPv4-IPv6 translation mechanism for SIP overlay network in UMTS all-IP environment

Both IPv6 and session initiation protocol (SIP) are default protocols for Universal Mobile Telecommunications System (UMTS) all-Internet protocol (IP) network. In the existing mobile telecommunications environments, an IPv6-based UMTS all-IP network needs to interwork with other Internet protocol version 4 (IPv4)-based SIP networks. Therefore, mobile SIP applications are typically offered through an overlay structure over the IPv4-Internet protocol version 6 (IPv6) interworking environments. Based on 3GPP 23.228, we propose an IPv4-IPv6 translation mechanism (i.e., SIPv6 translator) that integrates different IP infrastructures (i.e., IPv4 and IPv6) to provide an overlay network for transparent SIP application deployment. In this paper, we present the architecture and the call flows of the SIPv6 translator. An analytic model is proposed to investigate the fault tolerance issue of our approach. Our study provides guidelines to select appropriate number of processors for fault tolerance.

Performance measurements of TD-LTE, WiMax and 3G systems

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

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.

Effective VoIP call routing in WLAN and cellular integration

In cellular-WLAN integration, a dual-mode mobile station (MS) typically disables the WLAN module for power saving. A major problem is that for an incoming VoIP call (or data session), the MS will not be able to receive this call from the WLAN. It turns out that the call is directed to the cellular network. This letter proposes a simple push solution where an MS can accurately detect a VoIP call from paging signaling of the cellular network. Then the WLAN module of the MS is turned on and the VoIP call is connected to the MS through the relatively inexpensive WLAN.

NAT traversing solutions for SIP applications

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.