We study the performance of a statistical multiplexer whose inputs consist of a superposition of packetized voice sources and data. The performance analysis predicts voice packet delay distributions, which usually have a stringent requirement, as well as data packet delay distributions. The superposition is approximated by a correlated Markov modulated Poisson process (MMPP), which is chosen such that several of its statistical characteristics identically match those of the superposition. Matrix analytic methods are then used to evaluate system performance measures. In particular, we obtain moments of voice and data delay distributions and queue length distributions. We also obtain Laplace-Stieitjes transforms of the voice and data packet delay distributions, which are numerically inverted to evaluate tails of delay distributions. It is shown how the matrix analytic methodology can incorporate practical system considerations such as finite buffers and a class of overload control mechanisms discussed in the literature. Comparisons with simulation show the methods to be accurate. The numerical results for the tails of the voice packet delay distribution show the dramatic effect of traffic variability and correlations on performance.

A Markov Modulated Characterization of Packetized Voice and Data Traffic and Related Statistical Multiplexer Performance

The author studies a simple strategy, proposed independently by E.L. Hahne and R.G. Gallager (1986) and M.G.H. Katevenis (1987), for fairly allocating link capacity in a point-to-point packet network with virtual circuit routing. Each link offers its packet transmission slots to its user sessions by polling them in round-robin order. In addition, window flow control is used to prevent excessive packet queues at the network nodes. As the window size increases, the session throughput rates are shown to approach limits that are perfectly fair in the max-min sense. If each session has periodic input (perhaps with jitter) or has such heavy demand that packets are always waiting to enter the network, then a finite window size suffices to produce perfectly fair throughput rates. The results suggest that the transmission capacity not used by the small window session will be approximately fairly divided among the large window sessions. The focus is on the worst-case performance of round-robin scheduling with windows. >

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Round-robin scheduling for max-min fairness in data networks

We have studied the effects of random packet losses in digital speech systems based on 12-bit PCM and 4-bit adaptive DPCM coding. The effects are a function of packet length B and probability of packet loss P L . We have also studied tbe benefits of an odd-even sample-interpolation procedure that mitigates these effects (at the cost of increased decoding delay). The procedure is based on arranging a 2B -block of codewords into two B -sample packets, an odd-sample packet and an even-sample packet. If one of these packets is lost, the odd (or even) samples of the 2B -block are estimated from the even (or odd) samples by means of adaptive interpolation. Perceptual considerations indicate that packet lengths most robust to losses are in the range 16-32 ms, irrespective of whether interpolation is used or not. With these packet lengths, tolerable P L values, which are strictly input-speech-dependent, can be as high as 2 to 5 percent without interpolation and 5 to 10 percent with interpolation. These observations are based on a computer simulation with three sentence-length speech inputs, and on informal listening tests.

Effects of Packet Losses in Waveform Coded Speech and Improvements Due to an Odd-Even Sample-Interpolation Procedure

A novel multipoint communication paradigm, in which each destination receives a subset of the source's signal that corresponds to that destination's terminal and access bandwidth constraints, is presented. The approach to realizing this paradigm is based on integration of layered coding of the source's signal, routing based on bandwidth demand, optimization of signal parameters, and layered error control. The author gives an overview of several hierarchical signal coding techniques; presents methods for finding maximum bandwidth available to destination, establishing maximum-bandwidth routes; and optimally assigns bandwidth to the signal layers to maximize overall reception quality. Error control procedures whereby the network, source, and destinations cooperate to maintain layered-based data integrity, using erasure recovery coding and prioritized packet detection are also presented. >

Multipoint communication by hierarchically encoded data

This invention is suited for a packet network that carries bursty data. A mechanism is coupled to an edge node for coding bursty information into a digital format at either a first coding rate of a second coding rate which is less than the first rate. A mechanism senses traffic overload of one or more intermediate nodes carrying the bursty informaton. A mechanism which is responsive to the overload sensing mechanism causes the coding mechanism to switch from the first rate to the second rate while a traffic overload is sensed. Thus, network demand requirements are reduced and the overload will be alleviated.

Dynamic encoding rate control minimizes traffic congestion in a packet network

Integrated packet-switched networks have potential for providing improved performance by dynamically sharing transmission bandwidths between various users and user types, but new flow control methods are needed to deal with packetized voice traffic. This paper describes a packet voice flow control concept based on embedded speech coding. Results are presented from a computer simulation study of the technique in the context of a multilink wideband packet speech network. Several control methodologies are described, leading to an end-to end feedback approach that achieves stable operation and efficient utilization of network resources by adaptively matching transmitted voice bit rates to prevailing network conditions. Issues in the design of embedded speech coding algorithms are reviewed and a candidate structure based on channel vocoding principles is presented, along with the subjective results of some preliminary listening tests

A Technique for Adaptive Voice Flow Control in Integrated Packet Networks

Voice communication networks have traditionally been designed to provide either analog signal paths or fixed-rate synchronous digital connections between individual subscribers. These designs were aimed at accommodating the "streamlike" character of speech, which has traditionally been considered to flow from source to destination at a more or less constant rate. By way of contrast, interactive and computer-to-computer data transactions tend to be "bursty" in nature, and this has given rise to the development of packet-switching methods for data communications. The dichotomous nature of these two major traffic classes and the apparent conflict between the types of network services they require has resulted in the deployment of separate military communications facilities for voice and data. A challenge in the design of future systems is to achieve overall economy and flexibility in the allocation of resources via the efficient integration of both traffic types in common network facilities. This paper summarizes a number of advanced concepts for switching and flow control of combined voice and data traffic in integrated environments. Performance characteristics are described based on analysis results and computer simulation studies for both multilink terrestrial and broadcast satellite network topologies.

Voice Communication in Integrated Digital Voice and Data Networks

An all digital realization of a channel vocoder is described, including both computational algorithms and hardware structures. The machine, constructed almost entirely of digital elements, operates at a 2400 bit/s rate and produces synthetic speech, whose intelligibility and speaker recognition properties are comparable to those of an average telephone line. The extensive use of integrated digital circuitry allows size and cost reductions that are not currently possible in conventional high-quality analog vocoder systems.

T. Bially

Papers

A Technique for Adaptive Voice Flow Control in Integrated Packet Networks

Voice Communication in Integrated Digital Voice and Data Networks

A Digital Channel Vocoder