Admission policies for integrated voice and data traffic in CDMApacket radio networks

The authors derive optimal admission policies for integrated voice and data traffic in packet radio networks employing code division multiple access (CDMA) with direct-sequence spread spectrum (DS/SS) signaling. The network performance is measured in terms of the average blocking probability of voice calls and the average delay and packet loss probability of data messages. The admission scheme determines the number of newly arrived voice users that are accepted in the network so that the long-term blocking probability of voice calls is minimized. In addition, new data arrivals are rejected if the mean delay or the packet loss probability of data exceeds a desirable prespecified level. A semi-Markov decision process (SMDP) is used to model the system operation. Then, a value iteration algorithm is used to derive the optimal admission control. Two models for the other-user interference of the CDMA system are considered: one based on thresholds and another based on the graceful degradation of the CDMA system performance, and their performance is compared. These admission policies find application in emerging commercial CDMA packet radio networks including cellular networks, personal communication networks, and networks of LEO satellites for global communications     

Congestion control for multimedia services

The problem of congestion control in high-speed networks for multimedia traffic, such as voice and video, is considered. It is shown that the performance requirements of high-speed networks involve delay, delay-jitter, and packet loss. A framing congestion control strategy based on a packet admission policy at the edges of the network and on a service discipline called stop-and-go queuing at the switching nodes is described. This strategy provides bounded end-to-end delay and a small and controllable delay-jitter. The strategy is applicable to packet switching networks in general, including fixed cell length asynchronous transfer mode (ATM), as well as networks with variable-size packets     

Packetized voice/data integrated transmission on a token passingring local area network

The performance of a token-passing ring network with packetized voice/data mixed traffic is investigated through extensive simulations. Both data and voice users are modeled in the simulations. Data users produce bursty traffic. Voice traffic is modeled as having alternating talkspurts and silences, with generation of voice packets at a constant rate during talkspurts and no packet generation during silence periods. Token passing ring local area networks are shown to effectively handle both voice and data traffic. The effects of system parameters (e.g. voice packet length, talkspurt/silence lengths, data traffic intensity, and limited exhaustive service disciplines) on network performance are discussed     

Supplementary services in the H.323 IP telephony network

Traditionally, different networks were developed to handle voice, data, and video. The circuit-switched telephone network carried voice and the packet network carried data. Due to different deployment of these networks, different services were developed, such as voice mail in the telephone network and electronic mail on the Internet. With the revolution of multimedia in the computer industry, voice, video, and data are now being carried on both networks. Supplementary services, such as transfer and forwarding (which were originally developed for private telephone networks and later migrated to public telephone networks) are now being developed for packet networks. The standards for packet networks are being defined in the H.323-based series of ITU-T recommendations. This article provides the H.323 architecture for supplementary services, the differences in deployment of these services between the circuit-switched and packet-switched networks, and interworking of these services across hybrid networks     

Improving Voice‐Service Support in Cognitive Radio Networks

Voice service is very demanding in cognitive radio networks (CRNs). The available spectrum in a CRN for CR users varies owing to the presence of licensed users. On the other hand, voice packets are delay sensitive and can tolerate a limited amount of delay. This makes the support of voice traffic in a CRN a complicated task that can be achieved by devising necessary considerations regarding the various network functionalities. In this paper, the support of secondary voice users in a CRN is investigated. First, a novel packet scheduling scheme that can provide the required quality of service (QoS) to voice users is proposed. The proposed scheme utilizes the maximum packet transmission rate for secondary voice users by assigning each secondary user the channel with the best level of quality. Furthermore, an analytical framework developed for a performance analysis of the system, is described in which the effect of erroneous spectrum sensing on the performance of secondary voice users is also taken into account. The QoS parameters of secondary voice users, which were obtained analytically, are also detailed. The analytical results were verified through the simulation, and will provide helpful insight in supporting voice services in a CRN.     

Packet CDMA versus dynamic TDMA for multiple access in anintegrated voice/data PCN

A comparative evaluation of dynamic time-division multiple access (TDMA) and spread-spectrum packet code-division multiple access (CDMA) approaches to multiple access in an integrated voice/data personal communications network (PCN) environment are presented. After briefly outlining a cellular packet-switching architecture for voice/data PCN systems, dynamic TDMA and packet CDMA protocols appropriate for such traffic scenarios are described. Simulation-based network models which have been developed for performance evaluation of these competing access techniques are then outlined. These models are exercised with example integrated voice/data traffic models to obtain comparative system performance measures such as channel utilization, voice blocking probability, and data delay. Operating points based on typical performance constraints such as voice blocking probability 0.01 (for TDMA), voice packet loss rate 10^-3 (for CDMA), and data delay 250 ms are obtained, and results are presented     

Secured Self Organizing Network Architecture in Wireless Personal Networks

Secured self organizing network is an approach to computer network architecture that seeks to address the technical issues in heterogeneous networks that may lack continuous network connectivity. In delay tolerant network packets storage exists when there is any link breakage between the nodes in the network so delay is tolerable in this type of network during the data transmission. But this delay is not tolerable in wireless network for voice packet transmission. This evokes the use of wireless networks. In a network, different wireless network topologies are interoperating with each other so the communication across the network is called overlay network. This network is vulnerable to attacks due to mobile behaviour of nodes and frequent changes in topologies of the network. The attacks are wormhole attack and blackhole attack is analysed in this paper. They are critical threats to normal operation in wireless networks which results in the degradation of the network performance. The proposed recovery algorithm for wormhole and the isolation of blackhole will increase the performance of the network. The performance metrics such as throughput, packet delivery ratio, end–end delay and routing overhead of the network are evaluated.

     

A Technique for Adaptive Voice Flow Control in Integrated Packet Networks

Integrated packet-switched networks have potential for providing improved performance by dynamically sharing transmission bandwidths between various users and user types, but new flow control methods are needed to deal with packetized voice traffic. This paper describes a packet voice flow control concept based on embedded speech coding. Results are presented from a computer simulation study of the technique in the context of a multilink wideband packet speech network. Several control methodologies are described, leading to an end-to end feedback approach that achieves stable operation and efficient utilization of network resources by adaptively matching transmitted voice bit rates to prevailing network conditions. Issues in the design of embedded speech coding algorithms are reviewed and a candidate structure based on channel vocoding principles is presented, along with the subjective results of some preliminary listening tests     

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.     

Voice synchronization in packet switching networks

An algorithm for voice synchronization for packet switching networks is presented. The algorithm has been tested both in simulation and on a real network. The algorithm runs on the TRAME packet switching network for both the Vocoder and CELP DoD voice coding standards. Some results of these tests are presented. Some details of the algorithm development and implementation are given as well     

Study of information loss in packet voice systems

Once a voice buffer is full, it remains full for a certain period, during which many packets are possibly blocked, resulting in consecutive clippings in voice. The packet loss rate during this period changes slowly and has large fluctuations. It is shown that the temporal behavior of packet loss, especially at high rate, is inherently determined by voice correlation and system capacity and is independent of buffer size. Buffering may reduce the occurrence of short blocking periods associated with low rates packet loss but does not affect long ones associated with high packet loss rates. In fact, increasing the buffer size merely extends nonblocking periods, and thereby reduces the overall average packet loss rate, but packet-loss performance within existing blocking periods is not significantly improved. A simple tool is developed for calculating the boundary performance. It is found that it is possible to design a packet-switched voice system without buffering only at the expense of supporting a fewer number of calls. The issue of voice delay allocation between source and network is discussed, and it is shown that it is more effective to keep the network delay short while extending the source delay     

Multiclass rate control mechanism for prioritised voice service inpacket networks

A new multiclass rate control method for real-time voice traffic transmission in packet networks is proposed. The class for rate control is defined according to the requested voice quality. In the proposed control mechanism, a source coding rate is adjusted based on network feedback information and class type. It is shown that, using the proposed method, multiple voice qualities can be supported without voice quality degradation in traffic with low priority     

A dynamic rate control mechanism for source coded traffic in a fastpacket network

To achieve better statistical gain for voice and video traffic and to relieve congestion in fast packet networks, a dynamic rate control mechanism is proposed. An analytical model is developed to evaluate the performance of this control mechanism for voice traffic. The feedback delay for the source node to obtain the network congestion information is represented in the model. The study indicates that significant improvement in statistical gain can be realized for smaller capacity links (e.g., links that can accommodate less than 24 voice calls) with a reasonable feedback time (about 100 ms). The tradeoff for increasing the statistical gain is temporary degradation of voice quality to a lower rate. It is shown that whether the feedback delay is exponentially distributed or constant does not significantly affect performance in terms of fractional packet loss and average received coding rate. It is also shown that using the number of calls in talkspurt or the packet queue length as measures of congestion provides comparable performance     

MSR（ITU-T X.87/Y.1324）体系构架与系统特性

针对光城域网对综合业务的电信级需求，以城域网三网业务融合传送为主要领域，以基于弹性分组环RPR实现多业务传送为主要手段，研究了多业务传送与交换的简化方法，介绍了自主国际标准ITU-T X．87／Y．1324的协议要点、组网拓扑结构、传输架构、系统节点的组成和MSR通用帧格式。通过与RPR的比较，得出如下结论：MSR解决了语音、数据和视频等多业务(支路)分别在RPR各节点上，下的传送、支路组播、支路保护和性能监测问题。MSR成本较低，而且在同一网络平台上提供语音、数据和视频等支路业务需要这样的功能，它是目前基于分组的城域网多业务传送和出租的有效方法。     

Unified approaches to integrated-service networks–burst and fast packet switching

The increasing demand for communication services, coupled with recent technological advances in communication media and switching techniques, has resulted in a proliferation of new and expanded services. Currently, networks are needed which can transmit voice, data and video services in an application-independent fashion. Unified approaches employ a single switching technique across the entire network bandwidth, thus allowing services to be switched in an application-independent manner. This paper presents a taxonomy of integrated-service networks, including a look at NISDN, while focusing on unified approaches to integrated-service networks. The two most promising unified approaches are burst and fast packet switching. Burst switching is a circuit switching-based approach which allocates channel bandwidth to a connection only during the transmission of ‘bursts’ of information. Fast packet switching is a packet switching-based approach which can be characterized by very high transmission rates on network links and simple, hard-wired protocols which match the rapid channel speed of the network. Both approaches are being proposed as possible implementations for integrated-service networks. We survey these two approaches, and also examine the key performance issues found in fast packet switching. We then present the results of a simulation study of a fast packet switching network.     

Air interface switching and performance analysis for fast vertical handoff in cellular network and WLAN interworking

The integration of wireless local area network (WLAN) hotspot and the 3G cellular networks is imminently the future mode of public access networks. One of the key elements for the successful integration is vertical handoff between the two heterogeneous networks. Service disruption may occur during the vertical handoff because of the IP layer handoff activities, such as registration, binding update, routing table update, etc. In this paper, the network interface switching and registration process are proposed for the integrated WLAN/cellular network. Two types of fast vertical handoff protocols based on bicasting and non‐bicasting supporting real‐time traffic, such as voice over IP, are modeled. The performance of a bicasting based handoff scheme is analyzed and compared with that of fast handoff without bicasting. Numerical results and the simulation are given to show that packet loss rate can be reduced by the bicasting during handoff scheme without increasing bandwidth on both wireless interfaces. Copyright © 2006 John Wiley & Sons, Ltd.     

A reservation random-access protocol for voice/data integratedspread-spectrum multiple-access systems

Most code-division multiple-access (CDMA) systems described in the literature provide only one single service (voice or data) and employ the strategy of “one-code-for-one-terminal” for code-assignment. This assignment, though simple, fails to efficiently exploit the limited code resource encountered in practical situations. We present a new protocol called reservation-code multiple-access (RCMA), which allows all terminals to share a group of spreading codes on a contention basis and facilitates introducing voice/data integrated services into spread-spectrum systems. The RCMA protocol can be applied to short-range radio networks, and microcell mobile communications, and can be easily extended to wide area networks if the code-reuse technique is employed. In RCMA, a voice terminal can reserve a spreading code to transmit a multipacket talkspurt while a data terminal has to contend for a code for each packet transmission. The voice terminal will drop a long delayed packet while the data terminal just keeps it in the buffer. Therefore, two performance measures used to assess the proposed protocol are the voice packet dropping probability and the data packet average delay. Theoretical performance is derived by means of equilibrium point analysis (EPA) and is examined by extensive computer simulation     

Voice/Data Integration Using Circuit Switched Networks

This paper assesses the impact of integrating voice and data over circuit switched networks. Three main types of circuit switching are considered: 1) traditional circuit switching, 2)fast circuit switchingemploying advanced switching speeds, and 3) enhanced circuit switchingemploying time assigned speech interpolation (TASI) and adaptive data multiplexing (ADM) techniques. The circuit switching networks are evaluated in terms of two main network performance parameters: transmission efficiency and delay. In addition, an evaluation is made of such things as protocol and error control, precedence and preemption, routing and flow control, synchronization, voice continuity, probability of error or loss, and classmarking flexibility. One of the main conclusions of this paper is that circuit switching technologies have several deficiencies associated with providing integrated voice/data service and that the future lies in the effective use of packet and hybrid (circuit/packet) switching technologies.     

Voice packetization and compression in broadband ATM networks

Some methods of supporting voice in broadband ISDN, (B-ISDN) asynchronous transfer mode (ATM), including voice compression, are examined. Techniques for voice compression with variable-length packet format at DS1 transmission rate, e.g., wideband packet technology (WPT), have been successfully implemented utilizing embedded adaptive differential pulse code modulation (ADPCM) coding, digital speech interpolation (DSI), and block-dropping schemes. For supporting voice in B-ISDN, voice compression techniques are considered that are similar to those used in WPT but with different packetization and congestion control methods designed for the fixed-length ATM protocol at high speeds. Possible approaches for packetization and implementation of variable-bit-rate voice coding schemes are described. ADPCM and DSI for voice coding and compression and cell discarding (CD) for congestion control are considered. The advantages of voice compression and CD in broadband ATM networks are demonstrated in terms of transmission bandwidth savings and resiliency of the network during congestion     

Speech transmission performance planning in hybrid IP/SCN networks

The introduction of Internet protocol technology into traditional telecommunications networks is changing the nature of speech communication in those networks. As the current switched network infrastructure is augmented by packet networks, the needs of voice users of these hybrid networks must be given due consideration. These users are accustomed to high-quality connections with low distortion in the speech signal and low transmission delay in the speech path. In order for packetized networks to gain widespread acceptance for speech transmission services, it is necessary to maintain this high-quality performance in the evolving networks. We discuss relevant speech transmission performance requirements and the associated activities taking place in regional and international standards bodies