Channel de-allocation schemes for GSM/GPRS networks

Several channel de-allocation schemes for GSM/GPRS（General Packet Radio Service） networks are proposed in this paper. For DRA （Dynamical Resource Allocation） with de-allocation mechanism, if a new voice call arrives and finds that all the channels are busy, then one of the GPRS packets which occupy more than one channel for data transmission may release a channel for the new voice call. This paper presents 5 de-allocation mechanisms, i.e. DA-RANDOM, DA-RICHEST, DA-POOREST, DA-OLDEST and DA-YOUNGEST, to select the GPRS packet for releasing the appropriate channel. Simulation results show that DA-OLDEST achieves the best performance, especially in packets blocking probability, among all the de-allocation schemes. Although the performance of the proposed de-allocation schemes is not significantly different, they are all much better than that of the scheme without de-allocation.     

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.     

Wireless Internet access for mobile subscribers based on theGPRS/UMTS network

With the advent of IP technologies and the tremendous growth in data traffic, the wireless industry is evolving its core networks toward IP technology. Enabling wireless Internet access is one of the upcoming challenges for mobile radio network operators. The General Packet Radio Service is the packet-switched extension of GSM and was developed to facilitate access to IP-based services better than existing circuit-switched services provided by GSM. We illustrate how a visited mobile subscriber on a GPRS/UMTS network can access his/her home network via the gateway GPRS support node (GGSN). We also propose some implementation ideas on wireless Internet access for a remote mobile subscriber based on a GPRS/UMTS network     

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.     

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     

Channel allocation for GPRS

Based on the GSM radio architecture, the general packet radio service (GPRS) provides users data connections with variable data rates and high bandwidth efficiency. In the GPRS service, allocation of physical channels is flexible, i.e., multiple channels can be allocated to a user. We propose four algorithms for the GPRS radio resource allocation: fixed resource allocation (FRA), dynamic resource allocation (DRA), fixed resource allocation with queue capability (FRAQ), and dynamic resource allocation with queue capability (DRAQ). We develop analytic and simulation models to evaluate the performance for these resource allocation algorithms in terms of the acceptance rate of both GPRS packet data and GSM voice calls. Our study indicates that DRAQ (queuing for both new and handoff calls) outperforms other algorithms     

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.     

End-to-End Adaptive QoS Provisioning over GPRS Wireless Mobile Network

The General Packet Radio Service (GPRS) offers performance guaranteed packet data services to mobile users over wireless frequency-division duplex links with time division multiple access, and core packet data networks. This paper presents a dynamic adaptive guaranteed Quality-of-Service (QoS) provisioning scheme over GPRS wireless mobile links by proposing a guaranteed QoS media access control (GQ-MAC) protocol and an accompanying adaptive prioritized-handoff call admission control (AP-CAC) protocol to maintain GPRS QoS guarantees under the effect of mobile handoffs. The GQ-MAC protocol supports bounded channel access delay for delay-sensitive traffic, bounded packet loss probability for loss-sensitive traffic, and dynamic adaptive resource allocation for bursty traffic with peak bandwidth allocation adapted to the current queue length. The AP-CAC protocol provides dynamic adaptive prioritized admission by differentiating handoff requests with higher admission priorities over new calls via a dynamic multiple guard channels scheme, which dynamically adapts the capacity reserved for dealing with handoff requests based on the current traffic conditions in the neighboring radio cells. Integrated services (IntServ) QoS provisioning over the IP/ATM-based GPRS core network is realized over a multi-protocol label switching (MPLS) architecture, and mobility is supported over the core network via a novel mobile label-switching tree (MLST) architecture. End-to-end QoS provisioning over the GPRS wireless mobile network is realized by mapping between the IntServ and GPRS QoS requirements, and by extending the AP-CAC protocol from the wireless medium to the core network to provide a unified end-to-end admission control with dynamic adaptive admission priorities.     

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     

Radio resource management schemes for combined GSM/GPRS mobile systems

Data services like Web browsing, e‐mail and file transfer are becoming more and more popular in cellular systems. In contemporary systems like Global System for Mobile communications (GSM), data transfer has been circuit‐switched, that is, physical resources are allocated to a user for the entire call/session duration. However, this is inefficient in case of bursty traffic, where bursts are separated by long intervals of inactivity. This has been the main reason for the introduction of General Packet Radio Service (GPRS), which on the one hand acts as a mobile access network to the Internet, while on the other hand it enables the operator to offer a wide variety of value‐added services [Wireless Access Protocol (WAP) over GPRS, e/m‐banking, e/m‐commerce, push services, etc.] efficiently. However, in contemporary commercial implementations of GPRS the radio resource allocation algorithm does not take into account the Quality of Service (QoS)‐related service characteristics—although such information is exchanged between the terminal and the network—and consequently all service requests are treated the same way (‘best effort’). In this paper, we propose and evaluate via a simulation platform various Radio Resource Management (RRM) schemes capable of differentiating the handling of ‘service requests’ (in uplink and downlink), taking into account the GPRS‐related QoS parameters (precedence, reliability, delay, mean and peak throughput). The evaluation is performed for a range of voice (circuit‐switched) traffic loads, number of Transmit Receive eXchange (TRXs), offered data (packet‐switched) services characteristics, number of dedicated Packet Data Channels (PDCHs), and so on, taking into account the respective QoS requirements for both service types (circuit‐ and packet‐switched). Copyright © 2003 John Wiley & Sons, Ltd.     

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.     

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     

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.     

A push mechanism for GPRS supporting private IP addresses

The General Packet Radio Service (GPRS) provides a packet switched data service for mobile users. However, the existing GPRS specification does not provide the push mechanism and it is difficult to support GPRS services such as call termination (incoming call) for wireless voice over IP (VoIP). Based on the short message service, this paper proposes a push mechanism for GPRS supporting private IP addresses. Our approach does not need to modify the existing GPRS structure.     

Distributed rate adaptive packet access (DRAPA) for multicell wireless networks

Fueled by the explosive growth of the Internet, applications are demanding higher data rates and better services. Given the scarcity of radio resources, higher network capacities need to be achieved through more efficient use of the available bandwidth. Current cellular networks utilize frequency planning schemes that are optimized for circuit-switched applications, and thus is inherently problematic for future wireless packet networks with bursty, high peak-rate traffics. Random access schemes such as the ALOHA are seen as better solutions for packet networks. However, co-channel interference may significantly reduce the network throughput when the multicell load is heavy. In this paper, we propose a distributed rate adaptive packet access (DRAPA) scheme to combine the advantages of rate adaptation (in circuit-switched networks) and random access (in packet-switched networks). In particular, DRAPA allows terminal stations to transmit packets in random access fashion in the presence of brusty interference from neighboring cells. The packet code rate is adjusted according to interference level so that the retransmisson is controlled at an acceptable level. The DRAPA scheme subsumes two traditional schemes as the extreme cases, and has superior performance over the traditional schemes in terms of throughput and stability.     

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.     

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     

Queueing analysis for preemptive transmission in underlay CRNs

In cognitive radio applications, there are multiple different types of message queues. Existing queueing based works in underlay cognitive radio networks mainly focused on homogeneous packets. Therefore, we propose a new underlay scheme to deal with packets' heterogeneity by assuming high‐priority packets have preemptive transmission opportunities over low‐priority ones, and the former ones have impatient waiting time. We model the secondary user packets' behavior as a M/M/1+GI queueing system from the perspective of the whole system, which is represented by a two‐dimensional state transition graph. The reneging probability of high‐priority packets and the average waiting time in two‐class priority queues are analyzed. Then we extend the single‐channel model to multiple channels. Under this scenario, the closed form of low‐priority packets' expected system delay is derived. Simulation results demonstrate that the average waiting time of high‐priority packets decreases with the growing interference power threshold and the average waiting time of the low‐priority packet is proportional to the arrival rate of high‐priority packet and so on. This work may lay the foundation to design efficient Medium Access Control protocols and optimize long‐term system performance by carefully choosing system parameters. Copyright © 2015 John Wiley & Sons, Ltd.     

Performance analysis of a packet-switched synchronous voice/datatransmission system

The queuing behavior of voice and data traffic for a proposed voice/data multiplexing system is investigated. For synchronization of packetized voice in the packet-switched network, the master frame format of time-division multiplexing (TDM) is utilized. This scheme has discrete delay characteristics for packetized voice and provides a simple play-out method for voice signals. The performance of voice and data traffic is investigated by obtaining the cumulative distribution of delay of voice packets and the mean delay time of voice and data packets. The performances of the proposed system and the circuit-switched system are compared to time-assignment speech interpolation (TASI) in terms of the loss probability of voice traffic and the maximum allowable number of input voice channels for a given trunk capacity. The proposed system has been found to be more efficient than the circuit-switched system with TASI for integrating multiple voice channels in a single link. For a given trunk capacity, the system can accommodate about twice the number of input voice channels as the circuit-switched system     

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