Performance analysis of channel de-allocation schemes for dynamic resource allocation in GSM/GPRS networks

Channel de-allocation for GSM voice call (DASV) has been considered for dynamic resource allocation in GSM/GPRS networks. Two new de-allocation schemes are proposed: de-allocation for GPRS packet (DASP) and de-allocation for both GSM voice call and GPRS packet (DASVP). An analytic model with general GPRS data channel requirement is derived to evaluate the performance of the schemes in terms of GSM voice call incompletion probability, GPRS packet dropping probability, average GPRS packet transmission time and channel utilisation.     

GSM/GPRS分裂信道高效重分配策略

现有GSM/GPRS系统各种信道分裂策略都仅限于话音呼叫,即仅允许分裂信道分配给话音呼叫而不允许分配给包呼叫.因而,当系统没有空信道时,即使可以提供分裂信道,一个新到达的包呼叫也将被阻塞.本文提出的分裂信道重分配策略允许话音呼叫和包呼叫均可获得分裂信道,论文还研究了包呼叫得到分裂信道的条件.研究结果表明,允许将分裂信道分配给新到达包呼叫可以获得更优越的性能;在本文的参数设置下,在门限参数θ＝2/3时,新策略的话音阻塞率和平均包传输时间没有明显恶化,而包阻塞率和平均信道利用率却得以显著改善.     

Performance comparisons of dynamic resource allocation with/without channel de-allocationin GSM/GPRS networks

In this letter, we analyzed and compared the performance of dynamic resource allocation with/without channel de-allocation in GSM/GPRS networks. It is quite known that dynamic resource allocation allows communication systems to utilize their resources more efficiently than the traditional fixed allocation schemes. In GPRS, multiple channels may be allocated to a user to increase the transmission rate. In the case when there are no free channels in the system, some of these channels may be de-allocated to serve higher priority calls. The results show that with channel de-allocation mechanism, the voice blocking probability can be greatly reduced, especially at high GPRS traffic load. Besides, the scheme with channel de-allocation mechanism can achieve higher channel utilization.     

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.     

Performance evaluation of GSM/GPRS networks with channel re-allocation scheme

Previous work studying the channel allocation schemes in GSM/GPRS network commonly assume that one or two channels are required by a GPRS data for the sake of analytical simplicity. In this letter, we remove the assumption and generalize the GPRS data channel requirement (M channels). Additionally, we propose a channel re-allocation scheme (RAS), executed upon the channel release, by re-allocating the idle channels to the GPRS data which is currently using less than M channels. The example findings show that RAS can significantly decrease the voice call blocking probability and GPRS packet transmission time with slight channel utilization increase and negligible expense on GPRS packet blocking probability. Small M (e.g.M=2) will underestimate the performance achievements of the prior channel allocation scheme as well as RAS.     

Channel allocation for priority packets in the GPRS network

As the general packet radio service (GPRS) network begins to provide such as "push-to-talk" (PTT) service, delay-sensitive packets should be given higher priority in transmission. In this paper, we study two channel allocation schemes that implement priority queues for priority packets in the GPRS network: bitmap channel allocation (BCA) and uplink state flag channel allocation (USFCA). Our study shows that the transmission delay of priority packets in the GPRS network can be better guaranteed using USFCA.     

A Medium Access Control Protocol for Voice/Data Integrated Wireless CDMA Systems

In this paper, a medium access control protocol is proposed for integrated voice and data services in wireless local networks. Uplink channels for the proposed protocol are composed of time slots with multiple spreading codes per slot based on slotted code division multiple access (CDMA) systems. The proposed protocol uses spreading code sensing and reservation schemes. This protocol gives higher access priority to delay‐sensitive voice traffic than to data traffic. The voice terminal reserves an available spreading code to transmit multiple voice packets during a talkspurt. On the other hand, the data terminal transmits a packet without making a reservation over one of the available spreading codes that are not used by voice terminals. In this protocol, voice packets do not come into collision with data packets. The numerical results show that this protocol can increase the system capacity for voice service by applying the reservation scheme. The performance for data traffic will decrease in the case of high voice traffic load because of its low access priority. But it shows that the data traffic performance can be increased in proportion to the number of spreading codes.     

Efficient channel borrowing strategy for real-time services inmultimedia wireless networks

An efficient resource sharing strategy is proposed for multimedia wireless networks. We assume the channel resource in a wireless system is partitioned into two sets: one for voice calls and one for video calls. In the proposed channel borrowing strategy, voice calls can borrow channels from those pre-allocated to video calls temporarily when all voice channels are busy. A threshold type decision policy is designed such that the channel borrowing request will be granted only if the quality of service (QoS) requirement on video call blocking will not be violated during the duration of channel lending. An analytical model is constructed for evaluating the performance of the channel borrowing strategy in a simplified wireless system and is verified by computer simulations. We found that the proposed channel borrowing scheme can significantly reduce the voice call blocking probability while the increase in video call blocking probability is insignificant     

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.     

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     

Hybrid channel assignment scheme for accommodating voice/datatraffic in DS-CDMA cellular systems

A hybrid channel assignment (HCA) scheme in direct sequence-code division multiple access (DS-CDMA) systems for accommodating integrated voice/data traffic is proposed and the required power levels of voice and data traffic are derived. These levels can be used to maintain the minimum required link qualities of all calls. In the proposed scheme, delay-sensitive voice traffic is accommodated in circuit mode and delay-nonsensitive data traffic is accommodated in packet mode. The capacity region is derived and it can be used for controlling voice call admission and scheduling data packets. The proposed scheme can achieve a high link efficiency with reduced control overhead by statistically multiplexing voice and data traffic     

Scalable Schemes for Call Admission and Handover in Cellular Networks with Multiple Services

The introduction of new service categories withdifferent bandwidth requirements, e.g., data and multimedia, to cellularmobile radio networks makes many of the traditional mechanisms for controlingtraffic unusable orless efficient. The call admission and the handover handling are of the mostsensitive issues in this extension to new services. The performance of allservices includingthe traditional voice and the new services can be dramatically affected ifappropriate schemes are not used. In this paper, we propose call admission andhandover handling schemes for a cellular mobile network that offers twoservice types: voice and data. The data connections are assumed to transmitatdifferent transmission rates that are integer multiples to that of one radiochannel. In the case of congestion, the base station asks the active dataconnections to reduce their transmission rate in order to provide freechannels for the newly arrived request of both service types. This isbasically intended for incoming handover requests. The request will berejected if the transmission rate of the active connections reaches a givenminimum rate. Similar mechanism can also be used for new call arrivals, butsome priority can be given to handovers by setting a higher transmission ratethreshold for the new call rejection. As an extension to the proposedscalability, aqueuing of new calls is also proposed and analyzed. Analytical models werebuilt for the two proposed schemes together with the traditional channelreservation scheme. The effect of different traffic and configurationparameters on the performance measures like the grade of service, blockingprobabilities, and utilization, are studied using the proposed technique.Results show that the proposed schemes provide very good performance and morefairness among the different service types.     

异步光分组交换的缓存排序方案及性能

分析了简单的先到先服务(FCFS)光纤延迟线(FDL)循环占用方案性能,发现其分组丢失率(PLR)较高,提出3种输入分组按长度排序,寻找最小的FDL缓存优化分配方案.分析和仿真结果表明:约10%的分组排序后使3种方案都大大地能减小PLR,最小长度分组占用最小可用FDL缓存方案的性能最好.业务负载低于0.8时,排序的缓存方案对管理有限的FDL是有效的.     

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     

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.     

Evaluating the perceived voice quality on VoIP network using interpolated FIR filter algorithm

Voice over Internet Protocol (VoIP) is a popular communication service nowadays. VoIP reduces the cost of call transmission by passing voice and video packets through the available bandwidth for data packets through Internet protocol. The quality of the VoIP signal is degraded due to the various network impairments. The proposed scheme, interpolated finite impulse response, is implemented as post-processor after decoding the signal in VoIP system. The performance of the proposed scheme is evaluated for various network conditions. The results of the proposed scheme are measured with the objective measurement methods for signal quality evaluation. The performance of the proposed system is compared with the existing techniques for quality improvement in VoIP system. The results show much improvement in speech quality with the proposed scheme in comparison to other similar schemes.     

Delay sensitive scheduling schemes for heterogeneous QoS over wireless networks

Future wireless networks will support the growing demands of heterogeneous and delay sensitive applications. In this paper, a users' satisfaction factor (USF) is defined to quantify quality of service (QoS) for different types of services such as voice, data, and multimedia, as well as for different delay constraints. This USF not only predicts the final delivered QoS during transmission, but also take advantages of the fact that different packets can be decoded at different time in the receivers. Based on this USF, four types of scheduling schemes considering tradeoffs between system performance and individual fairness are proposed. These schemes explore the time, channel, and multi-user diversity to guarantee quality of service and enhance the network performance. From the simulation results, the proposed scheduling schemes achieve different tradeoffs between individual fairness and high system performance for the heterogeneous and delay sensitive applications, compared with the weighted round-robin and the modified proportional fairness scheduling schemes     

Efficiency of the GSM-GPRS Air Interface for Real-Time IP Traffic Flows With and Without Packet Dropping

One IP terminal can occupy a single slot or a multiple number of slots within time frames in the GSM and GPRS, respectively. A limited number of radio resources (slots) are allocated in a base station for such IP terminals. If one IP terminal can occupy only one slot discontinuously in a time frame, there is one possibility resorting to all IP terminals to preserve active mode at a time. Thus, the number of accepted call in the GSM is the same as that of the radio resource. Similarly, if one terminal can occupy a multiple number of slots discontinuously/dynamically in a time frame, the number of accepted calls is obtained by dividing the number of radio resources during that time by the maximum allowed number of slots per IP terminal. A burstiness factor is defined for the IP traffic over GSM-GPRS air interface. Traffic channel efficiency with a bursty real-time IP traffic is unacceptably low, especially with the range of acceptable call loss probabilities pertaining to a lower burstiness factor. The channel efficiency can be enhanced and the call loss probability can be suppressed significantly if a higher maximum number of calls is accepted. Allocated radio resources are less than the maximum number of packet transmissions at a time. Therefore, some packets could be dropped from the real-time transmission system. A complete analysis for the real-time IP packet transmission over the single slot GSM and dynamically variable multislot GPRS air interface without packet dropping, and with packet dropping that increases the channel efficiency is executed. Results show that the channel efficiency as well as the packet dropping probability increases with increasing call intensity, maximum number of admitted IP calls and the burstiness factor.     

Performance of Connectionless Data Service in High Density Microcell

Two categories of commercial wireless packet services exist today – a standalone system where the entire bandwidth is dedicated to the transmission of data packets (e.g. ARDIS, Mobitex), and an overlay system, where the unused channels of an existing cellular phone system are used (e.g. CDPD, GPRS). This paper examines the performance of the overlay service in a high density Manhattan street grid microcell. This type of cell is common in central urban areas. A simple analytical model is derived to describe the channel occupancy distribution in the microcell. The model is used to examine the performance of CDPD, a connectionless data service operating over a cellular system. Throughput and latency are evaluated under the two existing channel assignment schemes currently in use. A new assignment scheme is proposed, and is found to give better performance with minimal changes to the CDPD specification.