Techniques for Packet Voice Synchronization

Packet switching has been proposed as an effective technology for integrating voice and data in a single network. An important aspect of packet-switched voice is the reconstruction of a continuous stream of speech from the set of packets that arrive at the destination terminal, each of which may encounter a different amount of buffering delay in the packet network. The magnitude of the variation in delay may range from a few milliseconds in a local area network to hundreds of milliseconds in a long-haul packet voice and data network. This paper discusses several aspects of the packet voice synchronization problem, and techniques that can be used to address it. These techniques estimate in some way the delay encountered by each packet and use the delay estimate to determine how speech is reconstructed. The delay estimates produced by these techniques can be used in managing the flow of information in the packet network to improve overall performance. Interactions of packet voice synchronization techniques with other network design issues are also discussed.     

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.     

包交换系统中的一种DSI算法

该文将数字语音插空(DSI)技术应用于包交换系统中,给出了一种高效传输话音信号的实现方案。在话音信道中采用DSI技术可获得不少于两倍的插空增益,而各种包交换系统对信源速率变化的适应性又使得DSI的实现非常容易。多路话音信号经DAI处理后送入包交换系统传输可显著提高通信线路的利用率。理论分析和计算机模拟结果验证了该方案的良好性能。     

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.     

Coded Speech in Packet-Switched Networks: Models and Experiments

This paper considers the possibility of introducing packetized voice traffic into a packet-switched network. It is well known that the network must assure voice packets sufficient delay characteristics for conversational speech, i.e., low delay between speaker and listener and low delay jitter or variance. To reach these goals, simplified protocols and priority rules for voice handling are proposed and evaluated. A model of a packet switching node structure capable of handling both data and voice is derived for both analytical and simulation approaches. The use of low bit rate voice encoders is considered. The necessity of avoiding the transmission of silent intervals is discussed in relation to the behavior of packet voice receivers. Proposed strategies are compared by means of analytical tools and simulation experiments considering the presence of voice, interactive, and batch data packets.     

Paris: An approach to integrated high-speed private networks

This paper describes a design of a high-speed packet switching system for integrated voice, video and data communications. The system makes use of a simplified network architecture in order to achieve the low packet delay and high nodal throughput necessary for the transport of voice and video. A prototype of this system has been implemented and is now being tested under a variety of packet traffic loads. We have demonstrated that this system provides a cost-effective solution for private integrated networks.     

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     

A New Performance Measurement for Voice Transmission in Burst and Packet Switching

Voice transmission in burst switching is characterized by the process of talkspurt clipping, while in packet switching, it is characterized by the process of packet delay. In most analyses, the talkspurt clipping has been measured by the clipping probability averaged over all bits, and the packet delay has been measured by the delay performance averaged over all packets. The resulting measures overlook the duration of clipping in a talkspurt and the significant difference of delay in packets arriving at different times. Because of the nature of voice, different effects of these may result in substantially different degrees of voice distortion. This paper studies the worst case performance of both processes. The voice traffic is modeled as a process alternating between overload and underload periods. Statistically, more clipping and delay will be incurred while in the overload period. By worst case we mean that, in burst switching, we measure the worst case of talkspurt clipping duration in an overload period, while in packet switching, we measure the worst case of packet delay in an overload period. Furthermore, a simple closed form equation is derived which gives a very good approximation of the worst case mean packet delay performance. This equation can be more generally applied when the packet service time is to be geometrically distributed or when voice and data are to be integrated. The voice performances in burst switching and packet switching are also compared.     

A Distributed Experimental Communications System

The packet experimental communications system (packet XCS) is a new experimental voice and data switch. It uses a local-area network (LAN) for digital voice transmission, with local intelligence for switching. The packet XCS also has highly distributed control. The individual sites cooperate to provide user services as well as internal data management. We have learned that several local networks, including CSMA/CD networks, can be made to work well for voice transmission and that highly distributed control is practical in such a system. A system has been constructed which is used as a testbed for distributed voice and data communications experiments. This system is purely for experimentation and does not indicate a direction for future Bell System product offerings.     

Elastic basket switching

This paper proposes a new integrated switching system, ‘elastic basket switching’, for broadband and multimedia communications, including voice and high-speed data. In elastic basket switching (EBS), it is possible flexibly and efficiently to handle multimedia information by adaptively assigning communication resources according to communication requests and bandwidth of switched information. For continuous information, such as voice, EBS functions just as a circuit switching system, and for burst data it achieves high-efficiency bandwidth usage equivalent to a packet switching system by demand-assign type time-slot assignment. The detail of EBS and its application to a departmental system-orientated PBX are described. The traffic handling capability and details of the hardware structure are presented. The experimental system, including use of LSIs in the main parts of EBS is also described.