Performance evaluation of a CDMA protocol for voice and dataintegration in personal communication networks

Future wireless personal communication networks (PCN's) will require voice and data service integration on the radio link. The multiaccess capability of the code-division multiple-access (CDMA) technique has been widely investigated in the recent literature. The aim of this paper is to propose a CDMA-based protocol for joint voice and data transmissions in PCN's. The performance of such a protocol has been derived by means of an analytical approach both in terms of voice packet dropping probability and mean data packet delay. Voice traffic has been modeled as having alternated talkspurts and silences, with generation of voice packets at constant rate during talkspurts and no packet generation during silence gaps. A general arrival process is assumed for the data traffic. However, numerical results are derived in the case of a Poisson process. Simulation results are given to validate our analytical predictions. The main result derived here is that the proposed CDMA-based protocol efficiently handles both voice and data traffic. In particular, it is shown that the performance of the voice subsystem is independent of the data traffic     

A Channel Access Protocol for Integrated Voice/Data Applications

A new multiaccess protocol is proposed for an integrated voice/data application. The protocol, which is a variation of virtual time CSMA (VT-CSMA), takes advantage of the periodicity of voice packets and possesses a number of important features. With this protocol, voice stations appear to have a dedicated time-division multiplexed (TDM) slot, and the delay of a voice packet is bounded by the length of a frame (defined to be the period between two consecutive voice packets from a voice station). Also, the amount of data added to the channel has little effect on the voice traffic. When silence detection is used, many more voice conversations can be supported without losing the dedicated-slot characteristic. This is in contrast to a movingboundary TDM system where the excessive bandwidth saved by silence detection can only be used for data. The protocol requires no global synchronization and is easy to implement. Simulation results are presented to evaluate its performance.     

Integrating voice with the pi-persistentprotocol for unidirectional broadcast bus networks

A previously proposed protocol is extended to include voice for multiaccess communication over unidirectional broadcast bus networks. The authors use a framed approach for integrating these two traffic types where a frame, which is repeated, consists of a voice subframe followed by a data subframe. They use speech detectors for modeling speech sources in order to prevent the loss of about 60% of the bandwidth that would otherwise be required, and they use a moveable-boundary scheme in which the voice subframe size is estimated from the number of voice packets transmitted in the previous frame. The authors determine the fraction of speech loss and the fraction of wasted allocated bandwidth due to the estimator by formulating Markov chain for the number of ready voice stations at the frame boundaries. To analyze data performance, they use a nonhomogeneous Markov chain or a data station's buffer content just before the data slots visit that station. By proper control of the estimator function, they can guarantee an upper bound on the average speech loss at any voice station, and a combined voice-data throughput close to unity is achieved. Results from the analysis and from a simulation match closely     

Design issues relevant to developing an integrated voice/datamobile radio system

The various design issues related to developing an integrated voice/data mobile radio system, including high speed digital radio frequency modulation in a mobile environment, statistics for the talkspurt/silence gap composition of speech, switching schemes for voice/data integration, encoding techniques, and voice and data traffic statistics are discussed. A performance analysis is conducted for a typical design, showing that a voice-only mobile radio system can be upgraded to an integrated voice/data system capable of carrying the full voice and data loads without requiring additional radio channels and without compromising voice performance. Data traffic is only minimally delayed (46.2 ms mean delay) for a fully loaded 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     

支持话音/数据分组并传的UPMA多址接入协议

本文提出了一种新的、支持数据/话音业务并传的多址接入协议——根据用户数目妥善安排分组传输的多址接入(User-dependent Perfect-scheduling Multiple Access——UPMA)协议,它根据实际需求对上、下行带宽资源实行动态分配.UPMA协议对不同的业务类型赋予不同的优先级,并用轮询方式妥善地安排节点的分组传输;同时,它采用独特的帧结构,使话音业务总是能够得到优先传输.本文还提出了一种高效的竞争接入算法,以保证激活的节点能够快速接入信道.最后,对UPMA协议的性能进行了仿真并与MPRMA协议的性能进行了比较,结果证明UPMA协议有更好的性能.     

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     

Voice/data channel access integration in TDMA digital cellularnetworks

The integration of digital data capabilities in the soon to be widely deployed digital cellular networks, which were primarily designed for voice communications, offers a low-cost way to capture the large and ever growing market for mobile data services. The authors propose and evaluate a multiaccess protocol for integrating data traffic in the E(nhanced)-TDMA voice system with digital speech interpolation, which is an enhancement of the emerging North American digital cellular standard. The proposed, protocol combines random access with slot reservation mechanisms to statistically multiplex data packets with speech spurt packets over the shared terminal-to-base air channel. The integrated protocol requires no modification in the voice access protocol used in the E-TDMA system, and can attain performance close to that of an ideal voice/data multiplexer. Furthermore, the protocol may enable multislot assignment per TDMA frame to match the throughput needs of individual data terminals, and can accommodate application-dependent data transmission priorities     

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.     

C-PRMA: a centralized packet reservation multiple access for localwireless communications

Packet-switched technology has been demonstrated as effective in cellular radio systems with short propagation delay, not only for data, but also for voice transmission. In fact, packet voice can efficiently exploit speech on-off activity to improve bandwidth utilization over time division multiple access (TDMA). Such an approach has been first suggested in the packet reservation multiple-access (PRMA) technique, an adaptation of the reservation ALOHA protocol to the cellular environment. However, being PRMA-based on a fixed frame scheme, it cannot thoroughly take advantage of the very short propagation delays encountered in microcellular systems that allow, for instance, the immediate retransmission of packets lost because of the interference noise from adjacent cells. We present the centralized PRMA, a natural enhancement of PRMA, in which the base station (BS) plays a central role in scheduling the transmissions of mobile stations (MSs). As a consequence, the transmission scheduling is very flexible and can account for the different traffic rate and delay constraints that emerge from voice and data integration. A packet retransmission policy to recover corrupted packets can be implemented and operated efficiently to provide an acceptable grade of service, even in a very noisy environment. The simulation results presented show the quantitative improvements of the centralized packet reservation multiple-access (C-PRMA) performance with respect to PRMA     

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.     

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     

Evaluation of header compression schemes for IP-based wireless access systems

For mobile IP-based telephony (voice over IP) and IP-based real-time multimedia over cellular radio systems, an economically viable solution is needed. It is an absolute requirement that, for example, the 60-octet IPv6/UPD/RTP headers on IP telephony packets be reduced in size to conserve bandwidth in the radio spectrum. We evaluate the performance of two header compression schemes, based on RFCs 2508 and 3095, under the conditions of cellular radio access technology. The results presented in this article refer to voice and Web browsing traffic and are based on the implementation of compression/decompression algorithms for the aforementioned standards. We find that RFC 3095 performs significantly better if used for mobile communications.     

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     

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     

Block Reservation Time Division Multiple Access (BRTDMA) Protocol for a High Capacity Wireless Network

Next generation high capacity wireless networks need to support various types of traffic, including voice, video and data, each of which have different Quality of Service (QoS) requirements for successful transmission. This paper presents an advanced reservation packet access protocol BRTDMA (Block Reservation Time Division Multiple Access) that can accommodate voice and data traffic with equal efficiency in a wireless network. The proposed BRTDMA protocol has been designed to operate in a dynamic fashion by allocating resources according to the QoS criteria of voice and data traffic. Most of the existing reservation protocols offers reservation to voice traffic while data packets are transmitted using contention mode. In this paper we propose a block reservation technique to reserve transmission slots for data traffic for a short duration, which minimizes the speech packet loss and reduce the end-to-end delay for wireless data traffic. The optimum block reservation length for data traffic has been studied in a cellular mobile radio environment using a simulation model. Simulation results show that the BRTDMA protocol offers higher traffic capacity than standard PRMA protocol for integrated voice and data traffic and offers flexibility in accommodating multimedia traffic.     

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.     

Voice and data integration in the air-interface of a microcellularmobile communication system

A multiple-access protocol and a call acceptance algorithm for voice and data integration in a microcellular mobile communication system are presented. The protocol supports circuit-mode voice, burst-mode voice, and data. A hybrid multiplexing scheme with no boundaries performs statistical multiplexing, the call-level (for circuit-mode voice) and the talkspurt/message-level (for burst-mode voice and data). This scheme achieves high utilization of the available bandwidth compared to pure circuit switching, but with a lower quality in the latter two classes, due to delay during channel access on each talkspurt/message. A two-party transaction model for each class is implemented, giving a realistic load on uplink and downlink. A unified access procedure is presented, and the structure of the required control bursts is described. Performance is analyzed using simulation, and the optimum data-segment size is obtained. The maximum acceptable load is determined for various traffic mixes. A call acceptance algorithm is implemented, and typical simulation results for delay and call blocking are given     

An integrated CDMA code assignment model for voice and data traffic in wireless communications

Wireless personal communication requires a provision of integrated services of multimedia traffic, such as voice and data, over the radio link. The multiple access protocols of code-division multiple-access (CDMA) techniques have been widely investigated in the recent literature. This paper presents an innovative multiple access protocol for CDMA-based wireless communication systems by fully utilizing the characteristics of voice and data traffic. In other words, a voice terminal can reserve a spreading code to transmit packets in multiple talk spurts, while a data terminal can transmit packets by either using the unassigned codes or borrowing the codes from the voice terminals during their silent periods. We build mathematical models for voice and data subsystems, respectively. Two performance parameters, the average dropping probability for voice packets and the average transmission delay for data packets, are derived based on the equilibrium point analysis. The effects of the two performance parameters on the system performance are discussed by varying the code reservation intervals of the voice terminals.     

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.