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.     

Highly Efficient Voice—Data Integration over Medium and High Capacity Wireless TDMA Channels

A new medium access control (MAC) protocol for mobile wireless communications is presented and investigated. We explore, via an extensive simulation study, the performance of the protocol when integrating voice and data traffic over two wireless channels, one of medium capacity (referring mostly to outdoor microcellular environments) and one of high capacity (referring to an indoor microcellular environment). Data message arrivals are assumed to occur according to a Poisson process and to vary in length according to a geometric distribution. We evaluate the voice packet dropping probability and access delay, as well as the data packet access and data message transmission delays for various voice and data load conditions. By combining two novel ideas of ours with two useful ideas which have been proposed in other MAC schemes, we are able to remarkably improve the efficiency of a previously proposed MAC scheme [5], and obtain very high voice sources multiplexing results along with most satisfactory voice and data performance and quality of service (QoS) requirements servicing. Our two novel ideas are the sharing of certain request slots among voice and data terminals with priority given to voice, and the use of a fully dynamic low-voice-load mechanism.     

On Multiple Traffic Type Integration over Wireless TDMA Channels with Adjustable Request Bandwidth

A new medium access control (MAC) protocol for mobile wireless communications is presented and investigated. We explore, via an extensive simulation study, the performance of the protocol when integrating voice, video and data packet traffic over a wireless channel of high capacity (referring to an indoor microcellular environment). Depending on the number of video users admitted into the system, our protocol varies: a) the request bandwidth dedicated to resolving the voice users contention, and b) the probability with which the base station grants information slots to voice users, in order to preserve full priority for video traffic. We evaluate the maximum voice capacity and mean access delay, as well as the aggregate channel throughput, for various voice and video load conditions, and the maximum voice capacity, aggregate channel throughput and average data message delays, for various video, voice and data load conditions. As proven by the comparison with a recently introduced efficient MAC scheme (DPRMA), when integrating voice and video traffic our scheme obtains higher voice capacity and aggregate channel throughput. When integrating all three traffic types, our scheme achieves high aggregate channel throughput in all cases of traffic load.     

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 approximate analysis of TDMA out-of-slot random access protocols for microcellular mobile communications

In this paper, out-of-slot random access protocols for voice services that operate in microcellular environment are studied and simulated. The bearer service is assumed to be structured as time division multiple access/frequency division multiple access/frequency division duplex (TDMA/ FDMA/FDD). According to a stratification of information flow ascall, talkspurt, andpacket, the protocols are implemented at the talkspurt level. During a call, talkspurts generate a stream of packets. Each talkspurt has to reserve a voice time slot with a special control packet sent in a dedicate control slot (out of slot signaling). After a successful access, a voice slot is assigned for the duration of the talkspurt. This work concentrates on the out of slot random access method. When a transition from the idle state to the active state occurs, a voice terminal starts generating a talkspurt. Access for a voice slotV is then initiated via a dedicated control slotC. The time spent in gaining aV slot depends on the kind of random access protocol used in theC slots. Once the access reservation phase is successful, the talkspurt starts the second phase of information transmission in a freeV slot. If allV slots are occupied by other talkspurts, the new talkspurt is queued until aV slot becomes free. If the sum of the access and queueing times exceeds a thresh-old, a portion of the talkspurt is clipped. In our work we define an analytical model to evaluate the percentage of clipped voice packets. Simulations validate the analytical model.The second version of this work was rewritten while the author was a visiting scholar at WINLABThe IS-54 standard itself has the TDMA/FDMA structure. The ETDMA enhancement appears to be very much like what is described in this paper.     

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     

Voice over ATM using AAL2 and bit dropping: performance and calladmission control

Asynchronous transfer mode (ATM) adaptation layer 2 (AAL2) has been designed for efficient transport of voice, fax, and voiceband data (VBD) traffic over an ATM virtual circuit. The protocol helps achieve low latency and high bandwidth efficiency while applying suitable compression methods on voice/VBD/fax calls and silence elimination on voice calls. We analyze the performance and capacity of an ATM multiplexer based on AAL2 adaptation. We assume that embedded adaptive differential pulse code modulation (ADPCM) is used to compress voice, and silence elimination is used to achieve statistical multiplexing gain. The embedded ADPCM coding scheme allows selective dropping of less significant bits of voice during congestion in the ATM/AAL2 multiplexer. We compare the call capacities of voice multiplexers with and without bit dropping (BD). The performance models and results presented are based on fairly general assumptions and can be used for traffic engineering and call admission control in land-line or wireless ATM systems for a variety of voice/voiceband compression algorithms. A generalized algorithm for call admission control is also described     

Performance evaluation of packet data services over cellular voice networks

In this paper we develop a Markov chain modeling framework for throughput/delay analysis of data services over cellular voice networks, using the dynamic channel stealing method. Effective approximation techniques are also proposed and verified for simplification of modeling analysis. Our study identifies the average voice call holding time as the dominant factor to affect data delay performance. Especially in heavy load conditions, namely when the number of free voice channels becomes momentarily less, the data users will experience large network access delay in the range of several minutes or longer on average. The study also reveals that the data delay performance deteriorates as the number of voice channels increases at a fixed voice call blocking probability, due to increased voice trunking efficiency. We also examine the data performance improvement by using the priority data access scheme and speech silence detection technique.     

一种动态利用码资源的CDMA系统性能研究

本文提出了一种基于随机预约ALOHA访问方式,能支持话音和数据业务的动态使用码资源的码分多址访问协议。在该协议中,话音终端采用预约请求排队访问方式。数据终端采用时隙ALOHA方式传输数据分组。理论分析和计算机仿真结果表明,该协议能有效地提高系统码资源的利用率。在系统处于重负载情况下该协议能优先保证话音业务服务质量,而处于轻负载下系统码资源能为数据业务充分使用。     

Interaction of TCP and data access control in an integrated voice/data CDMA system

This paper considers the interaction between a proposed data access control scheme and the standardized error recovery schemes on the radio link of a voice/data CDMA system. A data access control scheme for combined voice-data CDMA systems has been proposed and studied in previous literature. The scheme aims to maintain a certain target voice signal to interference ratio (SIR); this is achieved by controlling the data load according to the measured voice SIR. The data users are allowed to transmit in a radio-link time slot with a certain permission probability, which is determined by the base station based on the measured voice SIR in the previous slot. As per the IS-99 standards, however, data transmission operates under the framework of TCP, which is a higher level end-to-end protocol. The TCP data unit, called a segment, is typically equivalent to several tens of physical layer frames; hence, a segment transmission takes up several tens of slots. Due to changes in the number of voice users in talkspurt (which occur on a time scale shorter than a segment transmission time), the slot level data access control scheme can introduce significant variability in the segment transmission time. The effect of such variability on the TCP timers, which operate at the segment level, is of interest. In this paper, an approximate upper bound on the data throughput, taking the presence of TCP into account, is computed. The results provide one with an insight into the interaction of the access control scheme with TCP; they also give practical pointers as to choosing suitable parameters and operating points for the scheme. This revised version was published online in June 2006 with corrections to the Cover Date.     

A Reliable Pipeline Protocol for the Message Service of a Land Mobile Satellite Experiment

This paper describes and analyzes a pipeline protocol for the data message communications of MSAT-X, a proposed experimental satellite-based mobile communications network. A demand-assigned multiple access protocol using pure ALOHA for making reservation requests has been developed for MSAT-X under error-free assumptions. Preliminary propagation studies indicate that the shortterm bit error rate of satellite channels in a mobile environment can be as high as 10^-3. Therefore, error-control schemes must be developed to ensure reliable transmissions. In this paper, we propose a retransmission scheme using selective repeat to minimize the end-to-end delay. We also use slotted ALOHA for making reservation requests to increase the overall system throughput. Since the number of channels available for reservation and data channels is essentially fixed for a given voice call blocking probability and a fixed call arrival rate, the analysis presented in this paper is also applicable to the integrated voice and data services of MSAT-X. Various operational scenarios have been investigated.     

Efficient silence suppression and call admission control through contention-free medium access for VoIP in WiFi networks

In view of the rapidly growing trend of migrating customers from traditional wired phones to mobile phones and then to VoIP services in the recent past, there is a tremendous demand for wireless technologies to support VoIP, specially on WiFi technologies which have already matured commercially. This has put forth great research challenges in the area of wireless VoIP. In this article we have addressed two core issues, efficient silence suppression and call admission control, in QoS provisioning for VoIP services in WiFi networks. In this connection we present a QoS-aware wireless MAC protocol called hybrid contention-free access (H-CFA) and a VoIP call admission control technique called the traffic stream admission control (TS-AC) algorithm. The H-CFA protocol is based on a novel idea that combines two contention-free wireless medium access approaches, round-robin polling and TDMA-like time slot assignment, and provides substantial multiplexing capacity gain through silence suppression of voice calls. The TS-AC algorithm ensures efficient admission control for consistent delay bound guarantees and further maximizes the capacity through exploiting the voice characteristic so that it can tolerate some level of non-consecutive packet loss. We expose the benefits of our schemes through numerical results obtained from simulations.     

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     

Queueing analysis of delay constrained voice traffic in a packetswitching system

Unlike data traffic, the voice packet stream from a node has very high correlation between consecutive packets. In addition, in order for the speech to be properly reconstructed, a delay constraint must be satisfied. A queueing model that accurately predicts packet loss probabilities for such a system is presented. Analytical results are obtained from an embedded bivariate Markov chain and are validated by a simulation program. Based on this model, the impact of the delay constraint, talkspurt detection thresholds, and packet size on packet loss are studied. Two schemes, named `instant' and `random', for discarding late packets are considered. Simulation results show that better performance can be obtained by using the latter scheme     

A novel self‐adaptive transmission scheme over an IEEE 802.11 WLAN for supporting multi‐service

The IEEE 802.11 wireless local area network (WLAN) media access control (MAC) specification is a hybrid protocol of random access and polling when both distributed coordination function (DCF) and point coordination function (PCF) are used. Data traffic is transmitted with the DCF, while voice transmission is carried out with the PCF. Based on the performance analysis of the MAC protocol for integrated data and voice transmission by simulation, this paper puts forward a self‐adaptive transmission scheme to support multi‐service over the IEEE 802.11 WLAN. The simulation results show that, on the premise of satisfying the maximum allowable delay of packet voice, the self‐adaptive transmission scheme can improve the data traffic performance and increase the WLAN capacity through dynamic and appropriate adjustment of the protocol parameters. Especially, voice traffic is sensitive to delay jitter, and the self‐adaptive scheme can effectively decrease it. Finally, it is worth noting that the adaptive scheme is easy to be realized, whereas no change in the MAC protocol is needed. Copyright © 2006 John Wiley & Sons, Ltd.     

Packet Reservation Window Multiple Access for Microcellular Voice/Data Transmission

We propose a new packet reservation multiple access (PRMA) scheme for the joint transmission of voice and data traffics in a microcellular medium. The collision resolution protocol within the system is based on a modification of the window random access algorithm, which has superior properties compared to the conventional slotted Aloha. The proposed algorithm, which we call packet reservation window multiple access (PRWMA), works in distinct modes for voice and data without prioritization, and the user performs slightly different operations depending on the information type. Simulation results show that PRWMA outperforms PRMA by a significant margin in terms of voice user capacity.     

Effects of base doping and carrier lifetime on differential current gain and temperature coefficient of 4H-SiC power bipolar junction transistors

Considering the circumstance of heterogeneous voice flows, first, by applying Markov chain, this paper proposes an unsaturated analytical model for the IEEE 802.11e EDCA protocol, which considers the condition of non-ideal transmission channel and the character of the occurrence of backoff countdown at the beginning of time slot in EDCA protocol. Furthermore, according to the proposed model, the media access delay and throughput of a flow are analysed, and the flow-oriented call admission control (CAC) scheme is proposed. Finally, the simulation results are shown to confirm that the proposed CAC scheme can guarantee the requirements of throughput and delay of voice flows, and can admit more voice flows to improve the utilisation efficiency of network resources by choosing the appropriate values of the minimum contention window or the appropriate varieties of voice flows.     

The RRA‐ISA multiple access protocol with and without simple priority schemes for real‐time and data traffic in wireless cellular systems

A multiple access protocol, based on a Reservation Random Access (RRA) scheme, is derived for a wireless cellular network carrying real-time and data traffic. Given a TDMA framed channel and a cellular structure, the aim of the protocol is that of maximizing the one-step throughput over an entire frame. This is achieved by deciding on the access rights at the cell base station, which then broadcasts this information at the beginning of the frame. The decision is made on the basis of binary channel feedback information (collision/no collision) over the previous frames, as well as of long term averages of packet generation rates at the mobile stations, assuming independence in the presence of packets at the latter. The resulting protocol has therefore been termed Independent Stations Algorithm (ISA), and the overall scheme RRA-ISA. As in other RRA protocols, time constrained (e.g., voice) traffic operates in a dynamic reservation mode, by contending for a slot in the frame with the first packet of a burst, and then keeping the eventually accessed slot for the duration of the burst; packets of the time constrained traffic unable to access a slot within a maximum delay are dropped from the input buffer. No such constraint is imposed on data traffic. Together with the “basic” version of the access algorithm, three other variants are presented, which exploit three simple different priority schemes in the RRA-ISA “basic” structure, in order to give a prominence to the voice service. The aim of these variants is to improve the performance in terms of the maximum number of stations acceptable in the system, by slightly increasing the data packets delay. All the proposed schemes are analyzed by simulation in the presence of voice and data traffic. Several comparisons show a relevant performance improvement (in terms of data delay and maximum number of voice stations acceptable within a cell) over other protocols that use ALOHA as a reservation mechanism (RRA-ALOHA or PRMA schemes). This revised version was published online in June 2006 with corrections to the Cover Date.     

Delay Related Issues in Integrated Voice and Data Networks

This paper focuses on network delays as they apply to voice traffic. First the nature of the delay problem is discussed and this is followed by a review of enhanced circuit, packet, and hybrid switching techniques: these include fast circuit switching (FCS), virtual circuit switching (VCS), buffered speech interpolation (SI), packetized virtual circuit (PVC), cut-through switching (CTS), composite packets, and various frame management strategies for hybrid switching. In particular, the concept of introducing delay to resolve contention in SI is emphasized, and when applied to both voice talkspurts and data messages, forms a basis for a relatively new approach to network design called transparent message switching (TMS). This approach and its potential performance advantages are reviewed in terms of packet structure, multiplexing scheme, network topology, and network protocols. The paper then deals more specifically with the impact of variable delays on voice traffic. In this regard the importance of generating and preserving appropriate length speech talkspurts in order to mitigate the effects of variable network delay is emphasized. The results indicate that a desirable length of talkspurt "hangover" of about 200 ms will accomplish this without unduly affecting speech activity, and that, under these circumstances, the perceptable threshold of variable talkspurt delay can be as high as about 200 ms average. As such, the results provide a useful guideline for integrated services system designers. Finally, suggestions are made for further studies on performance analysis and subjective evaluation of advanced integrated services systems.     

Dynamic Code Assignment Medium Access (DCAMA) Protocol for Wireless Integrated Services Networks

A medium access control (MAC) protocol for wireless mobile networks that supports integrated services and provides quality of service (QoS) support is presented and evaluated via simulation. A controlled random access protocol which allows all terminals to dynamically share a group of spread spectrum spreading codes is used. The protocol provides mobile terminals the access control required for efficient transfer of integrated traffic with QoS guarantees. Two service classes are provided; "best-effort" service, with priority queueing, and reserved bandwidth circuit service. The performance of the protocol is evaluated via simulation for traffic consisting of integrated voice, data and compressed video. The performance assessment measure is packet delay.