Handoff with DSP Support: Enabling Seamless Voice Communications across Heterogeneous Telephony Systems on Dual-Mode Mobile Devices

In this paper we investigate the problem of voice communications across heterogeneous telephony systems on dual-mode (WiFi and GSM) mobile devices. Since GSM is a circuit-switched telephony system, existing solutions that are based on packet-switched network protocols cannot be used. We show in this paper that an enabling technology for seamless voice communications across circuit-switched and packet-switched telephony systems is the support of digital signal processing (DSP) techniques during handoffs. To substantiate our argument, we start with a framework based on the Session Initiation Protocol (SIP) for vertical handoffs on dual-mode mobile devices. We then identify the key obstacle in achieving seamless handoffs across circuit-switched and packet-switched systems, and explain why DSP support is necessary in this context. We propose a solution that incorporates time alignment and time scaling algorithms during handoffs for supporting seamless voice communications across heterogeneous telephony systems. We conduct testbed experiments using a GSM-WiFi dual-mode notebook and evaluate the quality of speech when the call is migrated from WiFi to GSM networks. Evaluation results show that such a cross-disciplinary solution involving signal processing and networking can effectively support seamless voice communications across heterogeneous telephony systems.     

Design and performance analysis for data transmission in GSM/GPRSsystem with voice activity detection

To efficiently utilize the bandwidth of cellular mobile systems and offer service of high quality to both voice and data users, we propose a protocol to integrate packet-switched data traffic into current time-division multiple-access (TDMA)-type circuit-switched digital voice systems. We analyze the performance of the proposed system, which transmits data packets in the silent periods of a conversation with voice activity detection and adapts itself to the GSM/GPRS system, which uses the idle channels to provide data services. We show that the proposed protocol can increase the bandwidth utilization efficiency and improve the throughput/delay performance of the data transmission while minimizing the impact on the current GSM/GPRS service     

Packet-by-Packet Adaptive Scheme for Video Coding and Transmission Over Wireless IP Networks Using Rate-Compatible Punctured Turbo (RCPT) Codes

In this paper, we consider real-time video coding and transmission over packet-switched wireless IP networks, such as WLAN, using RCPT codes and joint source-channel coding (JSCC) with concentration on a packet-by-packet adaptive scheme. We present a systematic design methodology to enable the applicability of JSCC techniques. The performance of H.263+ video coding and transmission over wireless channel modeled as slow Rician fading channels using this approach is studied. Results indicate that a packet-by-packet adaptive RCPT-JSCC approach is of significant advantage for real-time video applications and leads to more acceptable video delivery quality over interference-limited and time-varying wireless networks.     

基于GPRS的IP电话技术研究

文章研究了一种的新的无线IP电话技术GPRS-VoIP,是一种基于GPRS接入的IP电话技术,可以实现和传统的基于电路交换的语音通话进行无缝切换.文中分析了该技术下的通话和传统GSM语音通话的无缝切换.文中还详细的分析了时廷、丢包、通话不连续等因素对基于GPRS接入的IP通话的影响和其对于带宽的需求,并提出了相应的解决方法.     

Transport of mobile application part signaling over Internetprotocol

The convergence of voice, data, and video networks is creating a new environment for telecommunications. In response to the changes, telecommunications equipment manufacturers and service providers are competing fiercely to bring an optimum solution to customers. The evolution of GSM to GPRS and to UMTS is a cellular wireless industry endeavour to meet this demand. This evolution will see the core wireless network infrastructure change from circuit-switched to packet-switched where voice and data are transported using IP as the common protocol. However, this poses a number of challenges, one of which is how to run the key mobile application part signaling protocols over IP. MAP defines the application protocols between switches and databases (e.g., MSC, VLR, SGSN, HLR) for supporting mobility management, security management, radio resource management, and mobile equipment management. UMTS supports both circuit-switched and packet-switched services     

Resource assignment in a fixed broadband wireless system

We propose a time slot reuse partitioning (TRSP) scheme for resource assignment in packet-switched fixed TDMA wireless networks. The TSRP scheme is based upon two key ideas. The first one is for the system to have more than one coexisting reuse pattern in the time domain. The second is to match a terminal or application to an appropriate reuse pattern, so as to guarantee the required quality of service (QoS). We define and then simulate a specific implementation of the TRSP scheme for an integrated voice and broadband data system. The results show that this scheme provides good performance for both voice and data     

Approach for voice quality and throughput estimation in wireless convergent networks

A common communications convergence scenario which is being adopted in personal communications relates to the combination of wireless and cellular networks by the use of multimode terminals. Since most of the wireless networks were initially dimensioned only for data communications, this paper shows how voice over wireless LAN dimensioning could be addressed under the optimal network throughput and the perspective of voice quality, using a simple approach. The maximum number of simultaneous users resulting from throughput is limited by the collisions taking place in the shared medium with the statistical contention protocol. The voice quality is conditioned by the delay and the packet loss in the contention protocol. Both approaches are analyzed within the scope of the voice codecs commonly used in voice over wireless LANs, to conclude that voice dimensioning based on network throughput and voice quality show complementary results. Additionally the use of low rate codecs in voice over wireless LANs is advantageous for the network performance point of view but may produce poor voice quality results. Mid range codecs like G729 could represent a trade-off for quality throughput. For these reasons, voice quality and wireless network throughput have to be taken into account in the network admission control, design and deployment to ensure a satisfactory user experience. The impact of handoff interval of wireless convergent networks on the conversation quality need also be assessed for a proper network design.     

On the radio capacity of TDMA and CDMA for broadband wireless packet communications

Studies of the capacity of cellular systems, stated in terms of the admissible number of remote users, have generally been limited to voice telephony. We address the problem of comparing the interference-limited performance of code-division multiple-access (CDMA) and time-division multiple-access (TDMA) systems in a packet-switched environment. The objective is to determine whether the capacity advantages claimed for circuit-switched CDMA still apply in a packet-switched environment, where the natural time diversity of bursty transmission may be a significant factor. Under a set of specific assumptions about the wireless environment (including path loss, shadow fading, multipath delay spread, cochannel interference, power control, and coding), we evaluate the number of users that can be admitted to the system while maintaining some desired quality-of-service (QoS) level. Four different classes of users with different characteristics and requirements are considered. The system capacity is found to depend significantly on the QoS objectives, which might be stated in terms of availability of some specified signal-to-interference level, packet-loss rate, or mean tolerable delay. The main finding is that strict requirements imposed on the radio access level tend to favor CDMA, whereas if some form of packet recovery is allowed at the higher layers (implying a relaxed set of requirements on the radio interface), then a somewhat higher capacity may be achieved by TDMA.     

Cross-layer resource allocation for integrated Voice/Data traffic in wireless cellular networks

A major task in next-generation wireless cellular networks is provisioning of quality of service (QoS) over the bandwidth limited and error-prone wireless link. In this paper, we propose a cross-layer design scheme to provide QoS for voice and data traffic in wireless cellular networks with differentiated services (DiffServ) backbone. The scheme combines the transport layer protocols and link layer resource allocation to both guarantee the QoS requirements in the transport layer and achieve efficient resource utilization in the link layer. Optimal resource allocation problems for voice and data flows are formulated to guarantee pre-specified QoS with minimal required resources. For integrated voice/data traffic in a cell, a hybrid time-division/code-division medium access control (MAC) scheme is presented to achieve efficient multiplexing. Theoretical analysis and simulation results demonstrate the effectiveness of the proposed cross-layer approach.     

Wireless mobile communications at the start of the 21st century

At the start of the 21st century, the wireless mobile markets are witnessing unprecedented growth fueled by an information explosion and a technology revolution. In the radio frequency arena, the trend is to move from narrowband to wideband with a family of standards tailored to a variety of application needs. Many enabling technologies including wideband code-division multiple access, software-defined radio, intelligent antennas, and digital processing devices are greatly improving the spectral efficiency of third-generation systems. In the mobile network area, the trend is to move from traditional circuit-switched systems to packet-switched programmable networks that integrate both voice and packet services, and eventually evolve toward an all-IP network. Furthermore, accompanied by wireless mobile location technology, wireless mobile Internet is expected to revolutionize the services that can be provided to consumers in the right place and at the right time. Wireless mobile communications may not only complement the well established wireline network; it may also become a serious competitor in years to come. We review the history of the wireless mobile communications, examine the current progress in standards and technologies, and discuss possible trends for wireless mobile solutions     

Real-time channel emulation for mobile communcation test beds

This article describes a methodology to model in real-time the channel quality variations for mobile communication systems. This method was used to implement a GSM hardware test bed employed to demonstrate the improved perceived voice quality obtained with link adaptation. Real-time operation is achieved by use of a large error pattern database, derived for time varying channel models. The modular design of the demonstrator and the reusability of the database enable a relatively straightforward extension to conduct novel multi-channel link level investigations. The proposed methodology can be adapted for a range of radio interfaces.     

Reservation arbitrated access for statistical multiplexing voicetraffic over dual-bus metropolitan area networks

Metropolitan area networks (MANs) are well suited to serve as broadband multiplexers for asynchronous transfer mode (ATM) networks, to facilitate enterprise networking and to support future wireless personal communication systems. We propose and analyze a novel reservation arbitrated (RA) access method which provides isochronous voice transport over dual-bus MANs while enabling statistical multiplexing among voice calls. In combination with a new cyclic capturing (CC) mechanism, RA access allows stations to capture and reserve isochronous voice channels in a fair and distributive manner. This paper presents the RA access protocol, derives an analytical model for general waste-free voice reservation protocols, and analyzes the performance of RA access by computer simulations validated by analytical calculations. To assess the actual voice quality, simulation results based on a real voice signal are also presented. Results indicate that RA access offers significant improvements in channel utilization, as compared to prearbitrated (PA) access, while providing an acceptable quality of service. Therefore, RA access offers an efficient voice transport mechanism for existing switched multimegabit data service (SMDS) networks employing the IEEE 802.6 protocol, as well as emerging ATM/MAN-based broadband networks     

乡村环境无线接入系统的设计

在农村电网自动化调度系统中,业务存在话音、电路数据和分组数据,是个典型的无线接入系统。其对于数据传输速率要求不是很高,但对数据的传输可靠度提出较高的要求。结合工程实际需求,提出采用扩频加时分机制构建了一个有中心的,一点对多点的无线接入的应用方案。基于对乡村环境传播特性的简述,提出了信道分配方案,详细介绍了分组数据可靠的传输的方法。     

Voice communications over zigbee networks

This article provides an overview of ZigBee-enabled wireless networks and discusses the feasibility of supporting voice communications over ZigBee networks. We begin by providing an overview of the ZigBee technology followed by an evaluation of voice quality and performance over such an impoverished wireless channel. Two types of voice communications, namely full-duplex voice over IP (VoIP) and half-duplex push-to-talk (PTT) are considered. Voice quality of VoIP is measured using the R-factor [1] (a well known objective speech quality metric). The quality of PTT, however, is evaluated based on packet-loss rate, delay, and jitter. The simulation results demonstrate that a low-power, low-rate wireless sensor network can support a limited range of voice services.     

A New Approach for Performance Evaluation of TCP over Interference-Limited Wireless Channels

A new metric for performance evaluation of transport control protocol(TCP) over wireless channels based on the interference-limited characteristics of code division multiple address(CDMA) system is proposed. According to the new metric, the performance of TCP over CDMA correlated channel for different protocol parameters and different versions is investigated. The results show that appropriate selection of protocol parameters and packet error rate(PER) operation point can improve significantly the capacity of packet-switched CDMA-based network.     

Smart antenna technologies for future wireless systems: trends and challenges

The adaptation of smart antenna techniques in future wireless systems is expected to have a significant impact on the efficient use of the spectrum, the minimization of the cost of establishing new wireless networks, the optimization of service quality, and realization of the transparent operation across multitechnology wireless networks. Nevertheless, its success relies on two considerations that have been often overlooked when investigating smart antenna technologies: first, the smart antennas features need to be considered early in the design phase of future systems (top-down compatibility); second, a realistic performance evaluation of smart antenna technique needs to be performed according to the critical parameters associated with future systems requirements (bottom-up feasibility). In this article an overview of the benefits of and most recent advances in smart antenna transceiver architecture is given first. Then the most important trends in the adoption of smart antennas in future system are presented, such as reconfigurability to varying channel propagation and network conditions, cross-layer optimization, and multi-user diversity, as well as challenges such as the design of a suitable simulation methodology and the accurate modeling of channel characteristics, interference, and implementation losses. Finally, market trends, future projections, and the expected financial impact of smart antenna systems deployment are discussed.     

Real-time voice over packet-switched networks

We discuss the architecture and technical viability of transporting real-time voice over packet-switched networks such as the Internet. The value of integrating voice and data networks onto a common platform is well known. The telephony industry has proposed the ATM standard as a means of upgrading the Internet to provide both real-time and data services. In contrast, voice services may be added to traditional IP networks that were originally designed for data transmission alone. We consider the feasibility and expected quality of service of audio applications over IP networks such as the Internet. In particular, we examine possible architectures for voice over IP and discuss measured Internet delay and loss characteristics     

Impact of Statistical Multiplexing on Voice Quality in Cellular Networks

This paper examines the quality of transmission of voice over cellular, packet-switched networks. The medium access mechanism in the uplink is simulated under various statistical multiplexing scenarios in order to assess the effect of front-end clipping on voice quality. Moreover, the simulation is implemented in a real-time demonstration platform utilized to acquire subjective indicators of voice quality by performing Mean Opinion Score (MOS) tests. Results from the MOS tests are reported, and an analysis of the obtained speech samples is presented. Finally, the results are summarized and potential further directions for the simulation tool and the speech models are discussed.     

Channel Allocation for GPRS with Buffering Mechanisms

General Packet Radio Service (GPRS) provides mobile users end-to-end packet-switched services by sharing the radio channels with voice and circuit-switched services. In such a system, radio resource allocation for circuit-switched and packet-switched services is an important issue, which may affect the QoS for both services significantly. In this paper, we propose two algorithms: Dynamic Resource Allocation with Voice and Packet queues (DRAVP) and Dynamic Resource Allocation with Packet and Voice queues (DRAPV) for channel allocation of the voice calls and packets. We propose analytic and simulation models to investigate the performance of DRAVP and DRAPV in terms of voice call incompletion probability, packet dropping probability, average voice call waiting time, and average packet waiting time. Our study indicates that the buffering mechanism for GPRS packets significantly increase the acceptance rate of GPRS packets at the cost of slightly degrading the performance of voice calls.