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.     

Design and performance evaluation of an RRA scheme for voice‐data channel access in outdoor microcellular environments

In PCS networks, the multiple access problem is characterized by spatially dispersed mobile source terminals sharing a radio channel connected to a fixed base station. In this paper, we design and evaluate a reservation random access (RRA) scheme that multiplexes voice traffic at the talkspurt level to efficiently integrate voice and data traffic in outdoor microcellular environments. The scheme involves partitioning the time frame into two request intervals (voice and data) and an information interval. Thus, any potential performance degradation caused by voice and data terminals competing for channel access is eliminated. We consider three random access algorithms for the transmission of voice request packets and one for the transmission of data request packets. We formulate an approximate Markov model and present analytical results for the steady state voice packet dropping probability, mean voice access delay and voice throughput. Simulations are used to investigate the steady state voice packet dropping distribution per talkspurt, and to illustrate preliminary voice-data integration considerations. This revised version was published online in June 2006 with corrections to the Cover Date.     

An Integrated Voice/Data System for VHF/UHF Mobile Radio

This paper presents the basic architecture and performance of a mobile radio multiaccess voice/data system. Natural pauses in conversational speech allow bandwidth saving through interleaving of data packets and talkspurts from different voice sources. A speech detector designed specifically for the mobile environment is presented. Blocking and delay performance of the multiaccess uplink is analyzed for voice traffic, assuming no traffic effects from the low priority data packets. Performance results from simulation are then presented for two downlink strategies in a two-hop virtual circuit in which a base station acts as a relay. The results verify also that the uplink analysis is valid for low voice traffic. For the data traffic, simulation results are presented in terms of data packet transmission delay and probability of collision with talkspurts. The results indicate that data flow may be limited by the collision factor. This work concludes that relative to conventional radio telephoning in which two channels are dedicated to each transmitter/receiver pair, a bandwidth reduction of 30-35 percent can be achieved.     

Traffic smoothing effects of bit dropping in a packet voicemultiplexer

The performance of a packet voice multiplexer queue in which the less significant bits of voiced packets are dropped during states of congestion in the multiplexer is examined. Using the results of simulation and analytical modeling, it is illustrated that bit dropping of voice packets significantly smooth the burstiness of superposition packet voice traffic by speeding up the packet service rate during critical periods of congestion in the queue. The smoothing effect renders it possible to approximate the superposition by a Poisson process for modeling a packet voice multiplexer with bit dropping. By comparison with a simulation, an analytical model based on the Poisson assumption is shown to produce quite accurate performance predictions. The results indicate that significant capacity and performance advantages are gained in the multiplexer as a result of the bit-dropping scheme     

A Cross-layer Dual Queue Approach for Improving TCP Fairness in Infrastructure WLANs

Fairness is one of the most important performance measures in IEEE 802.11 Wireless Local Area Networks (WLANs), where channel is accessed through competition. In this paper, we focus on the fairness problem between TCP uplink and downlink flows in infrastructure WLANs from the cross-layer perspective. First, we show that there exists a notable discrepancy between throughput of uplink flow and that of downlink flow, and discuss its root cause from the standpoint of different responses to TCP data packet drop and TCP ACK packet drop at the access point (AP) buffer. In order to mitigate this unfairness, we propose a dual queue scheme, which works in a cross-layer manner. It employs two separate queues at the AP, one for the data packets of downlink TCP flows and another for the ACK packets of uplink TCP flows, and selects these queues with appropriate probabilities so that TCP per-flow fairness is improved. Moreover, we analyze the behavior of the dual queue scheme and derive throughputs of uplink and downlink flows. Based on this analysis, we obtain the optimal queue selection probabilities for fairness. Extensive simulation results confirm that the proposed scheme is effective and useful in resolving the TCP unfairness problem without deteriorating overall utilization.     

Output and delay process analysis for slotted CDMA wirelesscommunication networks with integrated voice/data transmission

This paper presents the output and delay process analysis of integrated voice/data slotted code division multiple access (CDMA) network systems with random access protocol for packet radio communications. The system model consists of a finite number of users, and each user can be a source of both voice traffic and data traffic. The allocation of codes to voice calls is given priority over that to data packets, while an admission control, which restricts the maximum number of codes available to voice sources, is considered for voice traffic so as not to monopolize the resource. Such codes allocated exclusively to voice calls are called voice codes. In addition, the system monitoring can distinguish between silent and talkspurt periods of voice sources, so that users with data packets can use the voice codes for transmission if the voice sources are silent. A discrete-time Markov process is used to model the system operation, and an exact analysis is presented to derive the moment generating functions of the probability distributions for packet departures of both voice and data traffic and for the data packet delay. For some cases with different numbers of voice codes, numerical results display the correlation coefficient of the voice and data packet departures and the coefficient of variation of the data packet delay as well as average performance measures, such as the throughput, the average delay of data packets, and the average blocking probability of voice calls     

Performance of packet voice transmission using IEEE 802.16 protocol

The IEEE 802.16 standard defines three types of scheduling services for supporting real-time traffic, unsolicited grant service (UGS), real-time polling service (rtPS), and extended real-time polling service (ertPS). In the UGS service, the base station (BS) offers a fixed amount of bandwidth to a subscriber station (SS) periodically, and the SS does not have to make any explicit bandwidth requests. The bandwidth allocation in the rtPS service is updated periodically in the way that the BS periodically polls the SS, which makes a bandwidth request at the specified uplink time slots and receives a bandwidth grant in the following downlink subframe. In the ertPS service, the BS keeps offering the same amount of bandwidth to the SS unless explicitly requested by the SS. The SS makes a bandwidth request only if its required transmission rate changes. In this article we study the performance of voice packet transmissions and BS resource utilization using the three types of scheduling services in IEEE 802.16-based backhaul networks, where each SS forwards packets for a number of voice connections. Our results demonstrate that while the UGS service achieves the best latency performance, the rtPS service can more efficiently utilize the BS resource and flexibly trade-off between packet transmission performance and BS resource allocation efficiency; and appropriately choosing the MAC frame size is important in both the rtPS and ertPS services to reduce packet transmission delay and loss rate     

Joint admission/congestion control for wireless CDMA systemssupporting integrated services

A new data traffic control scheme is developed for maintaining the packet error rate (PER) of real-time voice traffic while allowing nonreal-time data traffic to utilize the residual channel capacity of the multi-access link in an integrated service wireless CDMA network. Due to the delay constraint of the voice service, voice users transmit their packets without incurring further delay once they are admitted to the system according to the admission control policy. Data traffic, however, is regulated at both the call level (i.e., admission control) and at the burst level (i.e., congestion control). The admission control rejects the data calls that will otherwise experience unduly long delay, whereas the congestion control ensures the PER of voice traffic being lower than a specified quality of service (QoS) requirement (e.g., 10 ^-2). System performance such as voice PER, voice-blocking probability, data throughput, delay, and blocking probability is evaluated by a Markovian model. Numerical results for a system with a Rician fading channel and DPSK modulation are presented to show the interplay between admission and congestion control, as well as how one can engineer the control parameters. The tradeoff of using multiple CDMA codes to reduce the transmission time of data messages is also investigated     

A demand assignment system for mobile users in a community ofinterest

The paper presents a high performance wireless access and switching system for interconnecting mobile users in a community of interest. Radio channel and time slot assignments are made on user demand, while the switch operations are controlled by a scheduling algorithm designed to maximize utilization of system resources and optimize performance. User requests and assignments are carried over a low-capacity control channel, while user information is transmitted over the traffic channels. The proposed system resolves both the multiple access and the switching problems and allows a direct connection between the mobile end users. The system also provides integration of voice and data traffic in both the access link and the switching equipment. The “movable boundary” approach is used to achieve dynamic sharing of the channel capacity between the voice calls and the data packets. Performance analysis based on a discrete time Markov model, carried out for the case of optimum scheduling yields call blocking probabilities and data packet delays. Performance results indicate that data packets may be routed via the exchange node with limited delays, even with heavy load of voice calls. Also the authors have proposed scheduling algorithms that may be used in implementing this system     

A Basic Dynamic Routing Problem and Diffusion

Diffusion theory has sometimes been successful in providing excellent approximate solutions to difficult queueing problems. Here we explore whether such methods can be used to analyze a basic dynamic routing strategy associated with a single idealized node in a data network. We analyze a dynamic routing policy where messages, or packets, that arrive at a certain node are routed to leave the node on the link having the shorter queue. In the model, message or packet arrivals are Poisson and the service time is exponentially distributed. We explore a heavy traffic diffusion method and we also discuss the limitations of an ad hoc approach to applying diffusion. For a node withKoutgoing queues we find, under the assumption of heavy traffic, the optimum dynamic strategy, in the sense of minimizing the average delay. When this optimum dynamic strategy is compared to a static strategy where the outgoing traffic is split among theKqueues, we find that the average delay for the dynamic system is better by a factor ofK.     

Performance analysis of DS/SSMA unslotted ALOHA system withvariable length data traffic

We analyze the throughput of a direct-sequence spread spectrum multiple access (DS/SSMA) unslotted ALOHA system with variable length data traffic. The system is analyzed for two cases: (1) systems without a channel load sensing protocol (CLSP) and (2) systems with a CLSP. The bit-error probability and the throughput are obtained as a function of the signal-to-noise ratio (SNR) during message transmission, considering the number of overlapped messages and the amount of time overlap. We assume that the generation of data messages is Poisson distributed and that the messages are divided into packets before transmission. The system is modeled as a Markov chain under the assumption that the number of packets in a message is geometrically distributed with a constant packet length. The throughput variance of the DS/SSMA unslotted ALOHA system with variable length data traffic is obtained as the Reed-Solomon code rate varies. Results show that a significant throughput improvement can be obtained by using an error-correcting code     

Dynamic Behavior of a CSMA-CD System with a Finite Population of Buffered Users

In packet broadcast networks with buffered users, queues of packets at the users interfere with each other through a shared broadcast channel. Therefore, it is difficult to analyze the performance of such Systems by classical queueing theory. This paper provides an approximate performance analysis of a slotted nonpersistent CSMA-CD system with a finite number of users, each having a buffer of finite capacity. We develop an approximate Markovian model of the system with a multidimensional state vector. To analyze the model, We utilize an approximate analytical technique called equilibrium point analysis (EPA). Then, the throughput and average packet delay characteristics are obtained and the system stability behavior is demonstrated. An approximate expression is also derived for the probability distribution of the number of packets in a buffer. Numerical results from both analysis and simulation are given to illustrate the accuracy of those analytic results. Using the analytic results, we examine the effect of the collision detection capability on the system performance. Furthermore, we consider how to select both the rescheduling probability and the buffer size that guarantee the system stability, keep the probability of buffer overflow sufficiently small, and at the same time minimize the average packet delay.     

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     

Performance of Integrated Packet Voice/Data Token-Passing Rings

This paper presents performance results that indicate that packetized voice service can be provided on a token-passing ring without adversely affecting the performance of data traffic. This is accomplished by introducing a relatively mild priority structure: stations are limited to a single packet transmission per medium access, and voice packets are given access priority over data packets at the same station. In addition, voice traffic is allowed longer packet lengths than data traffic. Several versions of this basic scheme are considered: 1) the number of active stations is constrained so that voice packets are guaranteed access within one packetization period, 2) no guarantee on access time is provided and voice packets are discarded when the waiting time exceeds one packetization period, and 3) no guarantee on access time is provided and voice packets are buffered until they can be transmitted.     

Incremental time-slot assignment in SS/TDMA satellite systems

The heterogeneous traffic in this environment can be categorized into a rapidly changing type composed of packet switched data traffic and a relatively static type composed of circuit switched voice traffic. From the time-slot assignment viewpoint, the problem is to construct an efficient TDMA frame that permits the static voice traffic to be transmitted and, then, on a frame-by-frame basis to attempt to insert the data packets into the slots that are unused by the voice traffic. It is proved that the problem is NP-complete, even for very simple traffic configurations. Several suboptimal fast heuristic algorithms are presented and empirically compared by experiments on randomly generated traffic patterns. The experiments reveal that, on the average, the algorithms give close to the optimal performance     

Analysis of Enhanced Collision Avoidance Scheme Proposed for IEEE 802.11e-Enhanced Distributed Channel Access Protocol

In enhanced distributed channel access (EDCA) protocol, small contention window (CW) sizes are used for frequent channel access by high-priority traffic (such as voice). But these small CW sizes, which may be suboptimal for a given network scenario, can introduce more packet collisions, and thereby, reduce overall throughput. This paper proposes enhanced collision avoidance (ECA) scheme for AC_VO access category queues present in EDCA protocol. The proposed ECA scheme alleviates intensive collisions between AC_VO queues to improve voice throughput under the same suboptimal yet necessary (small size) CW restrictions. The proposed ECA scheme is studied in detail using Markov chain numerical analysis and simulations carried out in NS-2 network simulator. The performance of ECA scheme is compared with original (legacy) EDCA protocol in both voice and multimedia scenarios. Also mixed scenarios containing legacy EDCA and ECA stations are presented to study their coexistence. Comparisons reveal that ECA scheme improves voice throughput performance without seriously degrading the throughput of other traffic types.     

Slot allocation for an integrated voice/data TDMA mobile radiosystem with a finite population of buffered users

Slot allocation for voice and data in an integrated TDMA mobile radio system is investigated. In the proposed system, voice traffic is circuit-switched and data traffic is packet-switched using slotted ALOHA for channel access; the data traffic model is practically assumed to have a finite number of users with finite buffer capacity. The authors apply an equilibrium point analysis (EPA) technique to analyze the data performance and present a heuristic performance criterion to obtain an optimal slot allocation for voice and data in the integrated TDMA mobile radio system     

An integrated voice/data system for mobile indoor radio networks

A centralized, integrated voice/data radio network for fading multipath indoor radio channels is proposed and analyzed. The packets of voice and data are integrated through a movable boundary method. The uplink channel access uses a framed-polling protocol whereas the downlink uses a time-division multiple-access (TDMA) scheme. This system dynamically switches between two transmission rates and uses multiple antennas to maximize the throughput in the fading multipath indoor environment. Throughput and delay characteristics of the system are analyzed using four different techniques. The results are compared with those of Monte Carlo computer simulations. A simple relationship between the number of voice terminals and the throughput of the data traffic are derived for an upper bound of 10-ms delay for the data packets     

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     

Admission control and buffer management of wireless communication systems with mobile stations and integrated voice and data services

This study presents models for management of voice and data traffic and new algorithms, which use call admission control as well as buffer management to optimise the performance of single channel systems such as wireless local area networks in the presence of mobile stations. Unlike existing studies, the new approach queues incoming voice packets as well as data packets, and uses a new pre-emption algorithm in order to keep the response time of voice requests at certain levels while the blocking of data requests is minimised. A new performance metric is introduced to provide uncorrelated handling of integrated services. Queueing related issues such as overall queue capacity, individual capacities for voice and data requests, the probability of blocking, and effects of waiting time on overall quality of service are considered in detail. Analytical models are presented and the results obtained from the analytical models were validated using discrete event simulations.