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     

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.     

A decomposition approximation for the analysis of voice/dataintegration

A closed-form decomposition approximation for finding the data performance in voice/data queuing systems is presented. The approximation is based on Courtois' (1977) decomposition/aggregation techniques and is applied to Senet hybrid multiplexing, movable boundary frame allocation schemes. The approximation is applied to both infinite and finite buffer systems. In the former case the approximation is valid only in the underload region and serves as an upper bound for the mean data queuing delay. In the finite buffer case it is valid for the whole data traffic range and is shown to improve as the number of channels increase, and deteriorates as the buffer size increases. For finite buffer systems upper and lower bounds for the decomposition approximation have also been derived. It is found that the lower bound is tight in the underload and low traffic region of the overload. In these same regions the decomposition approximation serves as a tight upper bound     

Capacity allocation for integrated voice/data transmission at apacket switched TOM

The integrated transmission of voice and data at a time-division multiplexer (TDM) is discussed and analyzed. The system operates in a frame format and the channel capacity is governed by the frame size. The allocation of channel capacity for the transmission of voice and data is performed by a controller. Digital speech interpolation (DSI) and embedded coding techniques are used to enhance the transmission efficiency and to facilitate the implementation of multiplexing. Using a dynamic programming approach, a capacity allocation policy which jointly optimizes the voice/data performance is introduced. Numerical results indicate that the aggregate throughput of the system can be improved with a slight degradation in voice quality     

A Model for Evaluating the Performance of an Integrated Circuit- and Packet-Switched Multiplex Structure

Recent papers have introduced a multiplexing structure for mixing voice and data traffic in an integrated telecommunications system. This structure utilizes a master frame format of a time division statistical multiplex facility. A certain portion of the frame is allocated to voice calls, and data traffic is assigned to the remaining frame capacity. To achieve a high transmission utilization, data are allowed to use any residual voice capacity momentarily available due to statistical variations in the voice traffic. The voice traffic is treated as a loss system and data packets are buffered. In this paper we derive exact analytical expressions for the key system perfomance measures, the probability of loss for voice calls, and the expected waiting time for data packets. Actually, two cases are considered, the one discussed above, called the movable boundary case, and one where the boundary is fixed; i.e., data are not allowed to utilize the residual voice capacity. The computational aspects of calculating actual numbers are discussed in some detail, and results are presented for typical cases.     

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.     

Design and Analysis of a Channel Assignment Scheme for CDMA/TDMA Mobile Networks

In this paper, a channel assignment scheme is proposed for use in CDMA/TDMA mobile networks carrying voice and data traffic. In each cell, three types of calls are assumed to compete for access to the limited number of available channels by the cell: new voice calls, handoff voice calls, and data calls. The scheme uses the movable boundary concept in both the code and time domains in order to guarantee the quality of service (QoS) requirements of each type. A traditional Markov analysis method is employed to evaluate the performance of the proposed scheme. Measures, namely, the new call blocking probability, the handoff call forced termination probability, the data call loss probability, the expected number of handoff and the handoff link maintenance probability are obtained from the analysis. The numerical results, which are validated by simulation, indicate that the scheme helps meet the QoS requirements of the different call types.     

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.     

Bounds on the mean system-size and delay for a movable-boundaryintegrated circuit and packet switched communications channel

A multiaccess communications channel which is shared on an integrated circuited-switched and packet-switched basis is considered. Applications include time-division multiplexing and demand-assigned/TDMA communication channels which are shared by circuit-switched services (such as isochronous voice) and packet-switched services (such as data). Frames of fixed duration are established and divided into two parts: one part is used for voice transmissions while the other part serves to accommodate packetized data traffic. These two parts are separated by a movable boundary, so that data traffic can use the frame capacity which is temporarily unoccupied by voice transmissions. An exact expression is derived for the mean packet queue-size and delay. To reduce the computational complexity involved in using this expression, simpler expressions for upper and lower bounds are derived for the mean packet queue-size and packet delay. Examples demonstrating the tightness of these numerically efficient bounds are presented     

Optimal fixed frame multiplexing in integrated line- and packet-switched communication networks

Recently, emphasis has been placed on integrated communication facilities capable of handling both line-switching and packet-switch-ing digital traffic. The problem of dynamically allocating the bandwidth of a trunk to both types of traffic is formulated as a Markovian decision process. Line switching is modeled as a time division multiplexing loss scheme over a varying portion of a fixed time frame. Packet-switching traffic is served through the remaining portion of the frame and requires queueing at the multiplexer-concentrator. Two different cost criteria are examined involving probability of blocking for line switching and average queueing delay for packets. The corresponding optimization problems are presented under reasonable simplifying assumptions. The movable boundary scheme suggested for commercial implementation of integrated multi-plexers is shown to offer optimal or near-optimal performance.     

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.     

A reservation protocol for packet voice and data integration inunidirectional bus networks

A movable boundary protocol is proposed for integrating packet voice and data in unidirectional bus networks. The head station on the bus learns the number of ready-to-transmit voice stations by reading a `request' bit in the header of the received packets and allocates the exact number of voice slots needed in each frame. The protocol guarantees that the maximum delay to transmit a voice packet will be less than the round-trip propagation delay at the head station plus twice the time needed to form the packet. The average data packet delay is evaluated via approximate analysis and simulation, for the case in which the voice-reserved slots in a frame are contiguous and for the case in which they are evenly distributed     

Voice/Data Integration Using Circuit Switched Networks

This paper assesses the impact of integrating voice and data over circuit switched networks. Three main types of circuit switching are considered: 1) traditional circuit switching, 2)fast circuit switchingemploying advanced switching speeds, and 3) enhanced circuit switchingemploying time assigned speech interpolation (TASI) and adaptive data multiplexing (ADM) techniques. The circuit switching networks are evaluated in terms of two main network performance parameters: transmission efficiency and delay. In addition, an evaluation is made of such things as protocol and error control, precedence and preemption, routing and flow control, synchronization, voice continuity, probability of error or loss, and classmarking flexibility. One of the main conclusions of this paper is that circuit switching technologies have several deficiencies associated with providing integrated voice/data service and that the future lies in the effective use of packet and hybrid (circuit/packet) switching technologies.     

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     

FLC-Based MBWIMA/UMTS Protocol for Multimedia Services in UTRA TDD Mode

An intelligent medium access control (MAC) protocol based on fuzzy logic control (FLC) is proposed and compared with a general packet radio system in UMTS (GPRS/UMTS), priority scheme and the movable boundary wireless integrated multiple access in UMTS (MBWIMA/UMTS) protocols. The integrated video/voice/data services of UMTS in UTRA TDD mode have different transmission properties. By fuzzy logic control, the resources of the wireless communication can be intelligent assigned for different types of mediums. The voice-video dropping probability and data packet delay are input to FLC to optimally select the maximum number of voice/video slots. Voice activity detector (VAD) and multiple access interference in single cell are also considered in the simulations.     

Performance evaluation of a modified PRMA protocol for joint voiceand data packet wireless networks

In this paper, a modified version of the packet reservation multiple-access (PRMA) protocol is proposed to provide spatially dispersed voice and data user terminals wireless access to a base station over a common short-range radio channel. An analytical approach is presented in order to derive system performance in terms of mean data message delay and voice packet dropping probability. A suitable permission probability design is also proposed to enhance system performance. Performance comparisons with an extension of the PRMA protocol to voice data systems previously reported in literature are shown to highlight the better behavior of this approach     

A reinforced movable boundary multiplexing scheme with adaptive interpolation: Modelling and performance evaluation

A novel multiplexing scheme for integrated networks characterized by the coexistence of circuitswitched and packet-switched traffic is described in this paper. The new scheme is realized by a reinforcement of the basic movable boundary hybrid-switching technique, by incorporating an adaptive interpolation within the circuit-switched subsystem. The adaptation mechanism is controlled by the level of congestion in the packet queue. A precise queueing model for the multiplexer is developed and an analytical evaluation of the key performance parameters of interest, namely the loss probability and expected delay of the packet-switched subsystem, and the freeze-out fraction of the circuit-switched subsystem, is conducted. The results of several numerical studies are presented to describe the performance of the multiplexer under different conditions. Of particular significance resulting from these studies is the trade-off between the freeze-out fraction of the circuit-switched subsystem and the loss probability of the packet-switched subsystem; this trade-off may be exploited in a systematic design of the interrated multiplexer tailored to specific applications.     

An efficient multiplexing technique for packet-switched voice/datanetworks

The authors propose a multiplexing frame structure that makes it possible to transmit voice messages synchronously without loss or clipping of contents. This scheme has discrete delay characteristics, and provides a simple play-out method for reproduction of voice signal. The authors investigate its performance by obtaining the cumulative distribution of delay of voice packets and the mean waiting time of data packets. It is concluded that this synchronous frame structure can easily be applied to enhance services with various transmission rates, such as flow control of message streams, node congestion control, and service-class or throughput-class negotiation of channels without significant degradation of trunk utilization     

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