As IP technologies providing both tremendous capacity and the ability to establish dynamic security associations between endpoints emerge, virtual private networks (VPNs) are going through dramatic growth. The number of endpoints per VPN is growing and the communication pattern between endpoints is becoming increasingly hard to predict. Consequently, users are demanding dependable, dynamic connectivity between endpoints, with the network expected to accommodate any traffic matrix, as long as the traffic to the endpoints does not overwhelm the capacity of the respective ingress and egress links. We propose a new service interface, termed a hose, to provide the appropriate performance abstraction. A hose is characterized by the aggregate traffic to and from one endpoint in the VPN to a set of other endpoints in the VPN, and by an associated performance guarantee. Hoses provide important advantages to a VPN customer: (1) flexibility to send traffic to a set of endpoints without having to specify the detailed traffic matrix, and (2) reduction in the size of access links through multiplexing gains obtained from the natural aggregation of the flows between endpoints. As compared with the conventional point-to-point (or customer pipe) model for managing quality of service (QoS), hoses provide reduction in the state information a customer must maintain. On the other hand, hoses would appear to increase the complexity of the already difficult problem of resource management to support QoS. To manage network resources in the face of this increased uncertainty, we consider both conventional statistical multiplexing techniques, and a new resizing technique based on online measurements. To study these performance issues, we run trace-driven simulations, using traffic derived from AT&T's voice network and from a large corporate data network. From the customer's perspective, we find that aggregation of traffic at the hose level provides significant multiplexing gains. From the provider's perspective, we find that the statistical multiplexing and resizing techniques deal effectively with uncertainties about the traffic, providing significant gains over the conventional alternative of a mesh of statically sized customer pipes between endpoints.

An increasing number of communication services are moving to an IP-based infrastructure. Packet telephony is probably the first important real-time service that must be supported well over an IP network. The use of IP presents a tremendous opportunity for service providers to exploit endpoint intelligence to offer creative new services going far beyond the current telephony service model. However, to support telephony, signaling protocols are needed that allow the service provider to offer network-layer service differentiation and, at the same time, to control access to both the enhanced network-layer quality of service as well as other services. This paper describes the distributed open signaling architecture (DOSA), which incorporates protocols that meet these needs. A key contribution of our work is a recognition of the need for coordination between call signaling, which controls access to telephony-specific services, and resource management, which controls access to network-layer resources.

As IP technologies providing both tremendous capacity and the ability to establish dynamic secure associations between endpoints emerge, Virtual Private Networks (VPNs) are going through dramatic growth. The number of endpoints per VPN is growing and the communication pattern between endpoints is becoming increasingly hard to forecast. Consequently, users are demanding dependable, dynamic connectivity between endpoints, with the network expected to accommodate any traffic matrix, as long as the traffic to the endpoints does not overwhelm the rates of the respective ingress and egress links. We propose a new service interface, termed a hose, to provide the appropriate performance abstraction. A hose is characterized by the aggregate traffic to and from one endpoint in the VPN to the set of other endpoints in the VPN, and by an associated performance guarantee.Hoses provide important advantages to a VPN customer: (i) flexibility to send traffic to a set of endpoints without having to specify the detailed traffic matrix, and (ii) reduction in the size of access links through multiplexing gains obtained from the natural aggregation of the flows between endpoints. As compared with the conventional point to point (or customer-pipe) model for managing QoS, hoses provide reduction in the state information a customer must maintain. On the other hand, hoses would appear to increase the complexity of the already difficult problem of resource management to support QoS. To manage network resources in the face of this increased uncertainty, we consider both conventional statistical multiplexing techniques, and a new resizing technique based on online measurements.To study these performance issues, we run trace driven simulations, using traffic derived from AT&T's voice network, and from a large corporate data network. From the customer's perspective, we find that aggregation of traffic at the hose level provides significant multiplexing gains. From the provider's perspective, we find that the statistical multiplexing and resizing techniques deal effectively with uncertainties about the traffic, providing significant gains over the conventional alternative of a mesh of statically sized customer-pipes between endpoints.

Packet telephony is of increasing interest in both the telecommunications and Internet communities. The emergence of packet telephony will create new services, and presents an opportunity to rethink how conventional telephony services are implemented. In this paper, we present an architecture for telephony over packet networks (TOPS). TOPS allows users to move between terminals or to use mobile terminals while being reachable by the same name. TOPS users can have multiple terminals and control how calls are routed to them. TOPS allows for terminals with a range of capabilities such as support for video, whiteboard, and other media with a variety of coding formats. TOPS retains the necessary information on terminal capabilities to determine the appropriate type of communication to be established with the remote terminal. The architecture assumes that the underlying network supports the establishment of end-to-end connectivity between terminals, with an appropriate quality of service. The components of TOPS are a directory service, an application layer signaling protocol, and a logical channel abstraction for communication between end-systems. The directory service maps a user's name to a set of terminals where the user may be reached. A user can control the translation operation by specifying profiles that customize how his name is mapped to a set of terminals where he can be reached. Terminal capabilities are also stored in the directory service. The application layer signaling protocol establishes and maintains call state between communicating terminals. The logical channel abstraction provides a shared end-to-end context for a call's constituent media and control streams, while isolating the applications from the details of the network transport mechanisms. In addition to supporting simple point-to-point calls, the architecture supports both centralized and decentralized conferencing. We also introduce a simple encapsulation format for voice.

We propose a scheme for transmission of variable bit rate (VBR) compressed video for interactive applications using the explicit-rate congestion-control mechanisms proposed for the available bit rate (ABR) service in asynchronous transfer mode networks. Compressed video is inherently bursty, with rate fluctuations over both short and long time scales. This source behavior can be accommodated by the ABR service, since the explicit-rate scheme allows sources to request varying amounts of bandwidth over time. Moreover, when the bandwidth demand cannot be met, the network provides feedback indicating the bandwidth currently available to a connection. In our scheme, the video source rate is matched to the available bandwidth by modifying the quantization level used during compression. We use trace-driven simulations to examine how effective the enhanced explicit-rate scheme is in "rate matching" between the network and the source and the effect on end-to-end delay. We also look at the sensitivity of the proposed scheme to the estimates of the network round-trip times and to inaccuracies in the rate requests made by sources.

The increasing deployment of wireless access technology, along with the emergence of high speed integrated service networks, such as ATM, promises to provide mobile users with ubiquitous access to multimedia information in the near future. One of the key problems in building connection-oriented ATM networks that support host mobility is designing mechanisms for rerouting virtual circuits to maintain data flow to and from mobile hosts. Ideally, VC rerouting must be done fast enough so as to cause minimal disruption to applications while minimizing the signaling overhead. In this paper, we evaluate the impact of several virtual circuit rerouting strategies on application performance. We initially identify the primitive operations required by any rerouting policy and use this to analytically quantify the cost of each rerouting policy in terms of wireless link disruption as a function of various network parameters. We then evaluate the effect of the VC rerouting policy on application-level performance using simulations. Our results show that the effect of rerouting policies are strongly dependent on the transport protocol policies and application QoS requirements, in addition to the network topology.

We propose a scheme for transmission of variable-bit-rate compressed video over ATM networks using the explicit-rate congestion control mechanisms proposed for the available bit rate (ABR) service. Compressed video is inherently bursty with rate fluctuations over both short and long time scales. We feel that this source behavior can naturally take advantage of the ABR service, since the ABR explicit-rate schemes allow sources to request varying amounts of bandwidth over time, while reserving a minimum for the entire duration of the connection. Moreover when the bandwidth demand cannot be met, the network provides feedback indicating the bandwidth currently available to a connection. This information can be used to match the video source rate to the available bandwidth by modifying the quantization level used during compression. We use trace driven simulations to examine how effective the enhanced explicit rate scheme is in "rate matching" between the network and the source and the effect on end-end delay. We also look at the sensitivity of the proposed scheme to the estimates of the network round-trip times and to inaccuracies in the rate requests made by sources.

There is an emergent interest in providing mobile users with ubiquitous wireless access to multimedia information. In this paper we present a network architecture and protocols to achieve this goal and describe their implementation in a prototype network called SWAN. Our network model assumes end to end ATM connectivity. Thus, the key question we address is how best to enhance ATM to support host mobility and wireless access.

The activity in building gigabit speed networks has led many researchers to re-examine the issue of congestion control. We describe a rate-based hop-by-hop congestion control mechanism in which the service rates of connections are dynamically adjusted at a switch, using feedback information provided by the neighboring switches. The desired service rate is computed based on a control equation that utilizes a model of the system with feedback information used to correct inaccuracies in the model. We use an analytical model to prove that the expected value of the queue occupancy and throughput of a controlled connection converge to the desired operating point. We also study the variation of the queue occupancy and throughput in steady-state as well as the transient response. The analytical results provide insights into how the parameter values chosen affect performance. We use simulations to compare the performance of the scheme with an equivalent end-to-end control scheme. Our analytical and simulation results show that the hop-by-hop scheme reacts faster to changes in the traffic intensity and, consequently, utilizes resources at the bottleneck better and loses fewer packets than the end-to-end scheme.

Future national and international telecommunication networks are headed towards using ATM (asynchronous transport multiplexing) technology. ATM technology allows seamless integration between different traffic types such as voice, video and data on a single network and also between customer premises' equipment and common carriers' networks. Unlike the current telephone networks, the ATM network provides virtual connections that have variable bandwidth and variable path delay between a sender and a receiver. Although ATM networks do support constant bandwidth channel as one of the traffic classes, using virtual connections as variable bandwidth pipes for video traffic is more attractive for reasons of efficiency in network usage and video quality. In this paper, we show that it is feasible to use congestion signals from networks as a feedback to the encoder and achieve very good perceptual quality for real-time video transmissions.

We show that modulating the source rate of a video encoder based on congestion signals from the network has two major benefits: the quality of the video transmission degrades gracefully when the network is congested and the transmission capacity is used efficiently. Source rate modulation techniques have been used in the past in designing fixed rate video encoders used over telephone networks. In such constant bit rate encoders, the source rate modulation is done using feedback information about the occupancy of a local buffer. Thus, the feedback information is available instantaneously to the encoder. In the scheme proposed, the feedback may be delayed by several frames because it comes from an intermediate switching node of a packet switched network. The paper shows the proposed scheme performs quite well despite this delay in feedback. We believe the use of such schemes will simplify the architecture used for supporting real time video services in future nationwide gigabit networks.

The authors examine the performance implications of providing reliability in conjunction with multicast transport over a high speed wide area network. They use a block based acknowledgement and selective retransmission protocol to evaluate the impact of the loss rate and the multicast tree topology on the achievable throughput. Their results show that even when the buffer overflow probability at switches and receivers is low, the cumulative loss probability seen by a source may be quite high. They also demonstrate that the average throughput increases significantly if the transport protocol delivers packets to the application layer out-of-sequence. They investigate the scaling properties of the error control mechanism and show that the multicast tree topology that results in minimum transfer time is not necessarily the same as the one constructed using minimal bandwidth or shortest path algorithms.>

A threshold-based dynamic allocation policy, called DQT, in which the bandwidth allocated to each class of traffic is altered based on the queue occupancy at the end of a frame is studied. Two classes of traffic with different quality of service (QOS) requirements are considered, and it is shown that the performance of the proposed scheme is better than that of a static allocation scheme. The performance of the scheme and the parameter sensitivity are studied under both stationary and nonstationary conditions, and it is indicated how parameter values can be chosen.>

In this paper we show that modulating the source rate of a video encoder based on feedback information from the network results in graceful degradation in picture quality during periods of congestion. Such source rate modulation techniques have been used in the past in designing video encoders used to generate data at a fixed rate. In such constant bit rate encoders, the source rate modulation is done using feedback information about the occupancy of a local buffer. Since the buffer is local, the feedback information is available instantaneously to the encoder. In the proposed scheme, the feedback information is delayed because it comes from within a packet-switching network. The feedback provides information about the traffic at switches along the path of the video connection. We show that the proposed scheme performs well, even though the information is delayed for relatively large intervals of time. We believe that the use of such schemes will simplify the architecture used for supporting real-time services in future nationwide gigabit networks.

The flow/congestion control scheme of TCP is based on the sliding window mechanism. As we demonstrate in this paper, the performance of this and other similar end-to-end flow control schemes deteriorates as networks move to the gigabit range. This has been the motivation for our search for a new flow and congestion control scheme. In this paper, we propose as an alternative, a hop-by-hop rate-based mechanism for congestion control. Due to the increasing sophistication in switch architectures, to provide “quality of service” guarantees for real-time as well as bursty data traffic, the implementation of hop-by-hop controls has become relatively inexpensive. A cost-effective implementation of the proposed scheme for a multi-gigabit packet switch is described in [2]. In this paper, we present results of a simulation study comparing the performance of this hop-by-hop flow control scheme to two end-to-end flow control schemes. The results indicate that the proposed scheme displays stable behavior for a wide range of traffic conditions and diverse network topologies. More importantly, the performance of the scheme, measured in terms of the average number of occupied buffers, the end-to-end throughput, the network delay, and the link utilization at the bottleneck, is better than that of the end-to-end control schemes studied here. These results present a convincing case against popular myths about hop-by-hop control mechanisms.