The author addresses state-of-the-art monitoring, surveillance, and operational techniques for maintaining digital transmission performance at required levels. Specific areas reviewed are the architectural framework in which operations and maintenance functions are carried out and the transport network elements to be maintained, including physical links and logical paths. Also presented are performance monitoring requirements which should be considered in the development of future industry standards. These requirements include digital transmission performance parameters and associated alert thresholds, as well as local memory requirements, and the modes of reporting performance information across a standard interface to a surveillance center.>

This paper addresses the problem of achieving overall quality-of-service of telecommunications networks in North America, following the divestiture of AT&T and the resulting competitive multivendor environment. The paper outlines an approach based on\bulleta top-down process in which quality-of-service is designed into networks by individual network vendors in cooperation with coordinating standards bodies, and\bulleta bottom-up process in which quality-of-service is verified and controlled under actual network operating conditions.

In this paper the listener echo performance of telephone network connections with more than one four-wire path is considered. The objective is to obtain simple design guidelines for the application of four-wire transmission and switching technologies, such that satisfactory listener echo performance for voice and voiceband data services is achieved. The performance analysis is based on a simple but accurate approximation of the overall connection frequency response. The amplitude response of each connection component is uniform. The general case of nonidentical four-wire paths and nonidentical interconnecting twowire paths is considered. Results are presented for: . the overall connection stability margin, and conditions for connection stability, . the condition for satisfactory connection performance, and . the relative merits of allocating required increases in connection stability margin to return losses, and to four-wire and two-wire path losses. A practical example illustrates the results.

The purpose of this paper is to examine speech impairments likely to arise in dynamically managed voice (DMV) systems. DMV systems utilize speech activity detection to exploit speech idle time and variable bit rate coding to exploit nonstationary speech statistics. The emphasis here is on systems using speech detection. This processing introduces two impairments not commonly found in traditional communication systems variable Speech burst delay and speech clipping. Simulations of these impairments were implemented, and formal subjective testing was performed to assess subjects' reactions to a range of impairment levels. Emphasis was on formal subjective listening tests and customer opinion of speech quality as defined by a rating scale. The test conditions are applicable to general telephony, where relatively high speech quality is required. Results on variable speech burst delay and front-end and midspeech burst clipping are presented. These results serve as input to the design process and to the establishment of performance guidelines for DMV systems.

This paper addresses top-down end-to-end user-oriented performance requirements pertaining primarily to voice and digital data services. The discussion of requirements for voice parameters accounts for the performance of existing analog and mixed analog/digital networks, as well as the likely effects on performance of short, medium, and long term evolution toward the ultimate all digital ISDN. The requirements for digital data parameters necessarily reflect an evolutionary process which is less consistent than for voice, and therefore these requirements are less definitive in nature. The discussions of voice and digital data performance apply largely to a wide variety of appropriate network designs, transmission schemes, and switching architectures. Both traditional parameters, as well as contemporary parameters associated with new and evolving systems, are considered. The emphasis is on the performance of nation-wide public and private networks, but the paper also considers the constraints of international connections.

This paper deals with the measurement and calculation of various speech temporal parameters of interest in an environment where speech activity detection is employed. In particular it is shown that, based on either a measurement or model of the probability density function (pdf) for silence durations for the case of zero talkspurt "hangover" or "fill-in," that the following temporal parameters can be computed for any value of hangover or fill-in: the mean (and pdf) for silence durations, the mean talkspurt duration, the mean talkspurt rate, and the speech activity. Directly measured values of these parameters and those computed from both measured and fitted versions of the pdf for silence durations are compared and are shown to be in reasonable agreement. The illustrated results are based on measurements of about two minutes of taped male monolog source speech. However, the approach to calculating the above parameters is general in the sense that it can be applied to any measured or modeled pdf for silence durations. The significance of this work lies in the important role that talkspurt hangover plays, for example, in minimizing speech detector induced back-end clipping of talkspurts, reducing exposure to the variable talkspurt delay impairment, and in determining signaling overhead and resource occupancy in various speech interpolation, packet voice, and integrated voice/data systems.

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.

This correspondence presents results related to measurmg the error performance of an experimental 12/14 GHz satellite link established between Carleton University, Ottawa, Canada, and NASA Ames Research Center, California, in 1976-1977. The measurements were performed on a 9.6 kbit/s voiceband data channel which supported the continuous synchronous transmission of fixed length frames (composite packets). The results include measurements of packet error rate versus uplink power settings under various atmospheric conditions, densities of packet error-free interval lengths, within-packet error statistics, and propagation delay variations.

This paper focuses on network delays as they apply to voice transmission. First the nature of the problem is discussed and this is followed by a review of circuit, packet and hybrid switching. The concept of introducing delay to resolve contention in speech interpolation is also noted, and when applied to both voice and data messages, forms a basis for a relatively new and so-called “transparent” approach to network design. The second part of the paper deals with experimental work. First, external link considerations involving subnet access by frame synchronous vocoders is discussed, and an experimental software interface is described which exploits the 60 to 65% idle time in voice calls. Secondly, based on the simulation of variable network delays, first indications are that with 180 ms “hangover”, the perceptable threshold of variable delay is about 200 ms average. Finally, suggestions for further works on performance analysis and subjective testing are made.

The paper describes the work done thus far in the development of a means of statistically multiplexing data and encoded voice in a transparent and intelligent network called TI-NET. A review of previous work in packetized voice transmission in a conventional packet switched network (ARPANET) has revealed problems related to large fixed and variable transmission delays. These problems result in degradations in speech quality in the form of time scale distortion and gaps due to very late or lost packets. The paper shows that TI-NET has various features which counter the above problems, and thereby make it appear very suitable for encoded voice transmission.The paper describes a software implementation of a protocol for encoded voice transmission which takes advantage of the 60 to 65% idle time in conversations (one way) so that only active periods in speech need be transmitted; this is not possible with present frame synchronous vocoders which transmit continuously.The paper describes the present experimental TI-NET which consists of two nodes (PDP11's) joined by a 9.6 kb/s link. For convenience the protocol for encoded voice transmission has been implemented in the TI-NET nodes, and the transparent transmission of data and encoded voice has been demonstrated. With regard to packetizing synchronous messages within TI-NET, a new protocol is described (and has been implemented) for padding partially filled “minipackets” that are likely to occur at the end of most synchronous messages.In addition, the paper discusses the features of another aspect of TI-NET, i.e. satellite extension nodes, which are used to enable both local and remote regions of high data concentration to access the subnet through 12-14 GHZ satellite links. It is shown that the advantages of accessing TI-NET by satellite extension nodes (as compared to accessing a conventional packet switched network by terrestrial facilities), include lower communication costs, greater accuracy and security, and smaller entrance delay.Finally, the paper describes an experiment in which data and encoded voice was transmitted from a TI-NET node at Carleton University, Ottawa, at 9.6 kb/s in “multiuser” packets, over the Hermes (CTS) satellite to NASA AMES Research Center in California, where it was looped back and returned to the same TI-NET node.The significance of the work described in the paper is that it represents an important step in the development of a transparent and intelligent public network capable of transmitting encoded voice (at 9.6 kb/s and lower rates) and data. Such a network can serve as an integrated system for data and voice and provide cost benefits to its users through savings in bandwidth of the order of 50 to 1, by statistically multiplexing data with vocoded voice.