Avoiding DAD for Improving Real-Time Communication in MIPv6 Environments

Joint International Workshops on Interactive Distributed Multimedia Systems and Protocols for Multimedia Systems, IDMS/PROMS 2002 Coimbra, Portugal, November 26–29, 2002 ProceedingsCurrent specification of address configuration mandates the execution of the Duplicate Address Detection (DAD) mechanism to prevent address duplication. However, a proper support for real time multimedia applications in mobile IPv6 nodes is undermined by the disruption imposed by DAD. In order to overcome this limitation, the usage of randomly generated IPv6 Interface Identifiers without previously performing DAD is proposed, based on the statistic uniqueness of the addresses generated through this method. The address duplication risk is quantified through the calculation of the probability of an Interface Identifier collision among the nodes sharing a link. The calculated probability is deemed to be negligible compared to other causes of communication failure, such as network outages.This research was supported by the LONG (Laboratories Over Next Generation Networks) project IST-1999-20393 and Moby Dick (Mobility and Differentiated Services in a Future IP Network) project IST-2000-25394

Quantification of audio quality loss after wireless transfer

The report describes a quality measurement for audio, both the theoretical background and implementation. It begins by describing the unlicensed methods the implementation is based on, Segmental SNR, Frequency Weighted Segmental SNR, Log-Likelihood Ratio, Cepstral Distance and Weighted Slope Spectral distance, and the commercial methods used as reference, PEAQ and PESQ. It also mentions the problems present in wireless transfer and the concept of sound quality assessment. It concludes by describing the suggested analysis method and implemented software together with the results when compared to PEAQ and PESQ.When talking on the phone, how do you know if the sound quality is good or bad? How do you know if it is better or worse than your last phone call? Although the perception of sound varies from person to person, only humans can truly determine sound quality. However, companies wants to ensure the quality of their product before releasing it, and therefore need an easier way to evaluate without humans, since human testing is expensive, time consuming and cannot be guaranteed to be consistent

On the synergy between adaptive physical layer and multiple-access control for integrated voice and data services in a cellular wireless network

In this paper, we propose a novel design to exploit the synergy between the multiple-access control (MAC) layer and the physical layer of a cellular wireless system with integrated voice and data services. As in a traditional design, the physical layer (channel encoder and modulator) is responsible for providing error protection for transmitting the packets over the hostile radio channel, while the MAC layer is responsible for allocating the precious bandwidth to the contending users for voice or data connections. However, a distinctive feature of our proposed design is that in the physical layer, a variable-rate adaptive channel encoder is employed to dynamically adjust the amount of forward error correction according to the time-varying wireless channel state such that the MAC layer, which is a reservation-based time-division multiple-access protocol, is able to make informed decisions as to bandwidth allocation. Specifically, based on the channel state information provided by the physical layer, the MAC protocol gives higher priority to users with better channel states. This novel synergistic mechanism between the two protocol layers can utilize the system bandwidth more effectively. The multiple-access performance of the proposed scheme is compared with two baseline systems. The first baseline system consists of the same reservation-based MAC protocol but with a traditional fixed-rate physical layer. The second system consists of the same reservation-based MAC protocol and the same channel adaptive physical layer, but without interaction between the two layers. All three protocols have a request queue, which stores the previous requests that survive the contention but are not allocated information slots. Our extensive simulation results demonstrate that significant performance gains are achieved through the exploitation of the synergy between the two protocol layers.published_or_final_versio

On channel-adaptive fair multiple access control

Multiple access control (MAC) of the uplink in a wireless mobile computing system is one of the most important resource allocation problems in that the response time and throughput of user applications (e.g., wireless Web surfing) are critically affected by the efficiency of the MAC protocol. Compared with a traditional MAC problem (e.g., the wireline Ethernet), there are two important new challenges in a modern wireless network: (1) multimedia data with diverse traffic requirements are involved; and (2) the wireless channel has a time-varying quality for each user. Furthermore, a more prominent user requirement is fairness among different users, possibly with different traffic demands. While some protocols have been suggested to handle multimedia data and/or tackling the time-varying channel, there are a number of drawbacks in these existing protocols. The most notable drawback is that the channel model is rather unrealistic - just using a two state Markov chain instead of relying on accurate models of multipath fading and shadowing effects. Another common deficiency is that fairness is ignored. In this paper, we propose to use a new notion of fairness that can capture a realistic channel model, and to integrate a fair queueing scheduling algorithm in a MAC protocol to optimize performance while maintaing fairness among users regardless of their channel states and data types.published_or_final_versio

Enhancement of perceived quality of service for voice over internet protocol systems

Voice over Internet Protocol (WIP) applications are becoming more and more popular in the telecommunication market. Packet switched V61P systems have many technical advantages over conventional Public Switched Telephone Network (PSTN), including its efficient and flexible use of the bandwidth, lower cost and enhanced security. However, due to the IP network's "Best Effort" nature, voice quality are not naturally guaranteed in the VoIP services. In fact, most current Vol]P services can not provide as good a voice quality as PSTN. IP Network impairments such as packet loss, delay and jitter affect perceived speech quality as do application layer impairment factors, such as codec rate and audio features. Current perceived Quality of Service (QoS) methods are mainly designed to be used in a PSTN/TDM environment and their performance in V6IP environment is unknown. It is a challenge to measure perceived speech quality correctly in V61P system and to enhance user perceived speech quality for VoIP system. The main goal of this project is to evaluate the accuracy of the existing ITU-T speech quality measurement method (Perceptual Evaluation of Speech Quality - PESQ) in mobile wireless systems in the context of V61P, and to develop novel and efficient methods to enhance the user perceived speech quality for emerging V61P services especially in mobile V61P environment. The main contributions of the thesis are threefold: (1) A new discovery of PESQ errors in mobile VoIP environment. A detailed investigation of PESQ performance in mobile VoIP environment was undertaken and included setting up a PESQ performance evaluation platform and testing over 1800 mobile-to-mobile and mobileto- PSTN calls over a period of three months. The accuracy issues of PESQ algorithm was investigated and main problems causing inaccurate PESQ score (improper time-alignment in the PESQ algorithm) were discovered . Calibration issues for a safe and proper PESQ testing in mobile environment were also discussed in the thesis. (2) A new, simple-to-use, V611Pjit ter buffer algorithm. This was developed and implemented in a commercial mobile handset. The algorithm, called "Play Late Algorithm", adaptively alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. It can be used as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance. Results show that the proposed algorithm can increase user perceived quality without consuming too much processing power when tested in live wireless VbIP networks. (3) A new QoS enhancement scheme. The new scheme combines the strengths of adaptive codec bit rate (i. e. AMR 8-modes bit rate) and speech priority marking (i. e. giving high priority for the beginning of a voiced segment). The results gathered on a simulation and emulation test platform shows that the combined method provides a better user perceived speech quality than separate adaptive sender bit rate or packet priority marking methods

The good, the bad, and the muffled: The impact of different degradations on Internet speech

This paper presents an experiment comparing the relative impact of different types of degradation on subjective quality ratings of interactive speech transmitted over packet-switched networks. The experiment was inspired by observations made during a large-scale, long-term field trial of multicast conferencing. We observed that user reports of unsatisfactory speech quality were rarely due to network effects such as packet loss and jitter. A subsequent analysis of conference recordings found that in most cases, the impairment was caused by end-system hardware, equipment setup or user behavior. The results from the experiment confirm that the effects of volume differences, echo and bad microphones are rated worse than the level of packet loss most users are likely to experience on the Internet today, provided that a simple repair mechanism is used. Consequently, anyone designing or deploying network speech applications and services ought to consider the addition of diagnostics and tutorials to ensure acceptable speech quality

A novel channel-adaptive uplink access control protocol for nomadic computing

We consider the uplink access control problem in a mobile nomadic computing system, which is based on a cellular phone network in that a user can use the mobile device to transmit voice or file data. This resource management problem is important because an efficient solution to uplink access control is critical for supporting a large user population with a reasonable level of quality of service (QoS). While there are a number of recently proposed protocols for uplink access control, these protocols possess a common drawback in that they do not adapt well to the burst error properties, which are inevitable in using wireless communication channels. In this paper, we propose a novel TDMA-based uplink access protocol, which employs a channel state dependent allocation strategy. Our protocol is motivated by two observations: 1) when channel state is bad, the throughput is low due to the large amount of FEC (forward error correction) or excessive ARQ (automatic repeated request) that is needed and 2) because of item 1, much of the mobile device's energy is wasted. The proposed protocol works closely with the underlying physical layer in that, through observing the channel state information (CSI) of each mobile device, the MAC protocol first segregates a set of users with good CSI from requests gathered in the request contention phase of an uplink frame. The protocol then judiciously allocates channel bandwidth to contending users based on their channel conditions. Simulation results indicate that the proposed protocol considerably outperforms five state-of-the-art protocols in terms of packet loss, delay, and throughput.published_or_final_versio

A performance study of multiple access control protocols for wireless multimedia services

The multiple access control (MAC) problem in a wireless network has intrigued researchers for years. For a broadband wireless multimedia network such as wireless ATM, an effective MAC protocol is very much desired because efficient allocation of channel bandwidth is imperative in accommodating a large user population with satisfactory quality of service. Indeed, MAC protocols for a wireless ATM network, in which user traffic requirements are highly heterogeneous (classified into CBR, VBR, and ABR), are even more intricate to design. Considerable research efforts expended in tackling the problem have resulted in a myriad of MAC protocols. While each protocol is individually shown to be effective by the respective designers, it is unclear how these different protocols compare against each other on a unified basis. We quantitatively compare seven previously proposed TDMA-based MAC protocols for integrated wireless data and voice services. We first propose a taxonomy of TDMA-based protocols, from which we carefully select seven protocols, namely SCAMA, DTDMA/VR, DTDMA/PR, D4RUMA, DPRMA, DSA++, and PRMA/DA, such that they are devised based on rather orthogonal design philosophies. The objective of our comparison is to highlight the merits and demerits of different protocol designs.published_or_final_versio

A quantitative comparison of multiple access control protocols for wireless ATM

The multiple access control (MAC) problem in a wireless network has intrigued researchers for years. For a broad-band wireless network such as wireless ATM, an effective MAC protocol is very much desired because efficient allocation of channel bandwidth is imperative in accommodating a large user population with satisfactory quality of service. Indeed, MAC protocols for a wireless ATM network in which user traffic requirements are highly heterogeneous (classified into CBR, VBR, and ABR), are even more intricate to design. Considerable research efforts expended in tackling the problem have resulted in a myriad of MAC protocols. While each protocol is individually shown to be effective by the respective designers, it is unclear how these different protocols compare against each other on a unified basis. In this paper, we quantitatively compare seven recently proposed TDMA-based MAC protocols for integrated wireless data and voice services. We first propose a taxonomy of TDMA-based protocols, from which we carefully select seven protocols, namely SCAMA, DTDMA/VR, DTDMA/PR, DQRUMA, DPRMA, DSA++, and PRMA/DA, such that they are devised based on rather orthogonal design philosophies. The objective of our comparison is to highlight the merits and demerits of different protocol designs.published_or_final_versio