Fouad A. Tobagi

Packet Switching in Radio Channels: Part I--Carrier Sense Multiple-Access Modes and Their Throughput-Delay Characteristics

Radio communication is considered as a method for providing remote terminal access to computers. Digital byte streams from each terminal are partitioned into packets (blocks) and transmitted in a burst mode over a shared radio channel. When many terminals operate in this fashion, transmissions may conflict with and destroy each other. A means for controlling this is for the terminal to sense the presence of other transmissions; this leads to a new method for multiplexing in a packet radio environment: carrier sense multiple access (CSMA). Two protocols are described for CSMA and their throughput-delay characteristics are given. These results show the large advantage CSMA provides as compared to the random ALOHA access modes.

Fast packet switch architectures for broadband integrated services digital networks

Background information on networking and switching is provided, and the various architectures that have been considered for fast packet switches are described. The focus is solely on switches designed to be implemented electronically. A set of definitions and a brief description of the functionality required of fast packet switches are given. Three basic types of packet switches are identified: the shared-memory, shared-medium, and space-division types. Each of these is described, and examples are given. >

Multiaccess Protocols in Packet Communication Systems

The need for multiaccess protocols arises whenever a resource is shared by many independent contending users. Two major factors contribute to such a situation: the need to share expensive resources in order to achieve their efficient utilization, or the need to provide a high degree of connectivity for communication among independent subscribers (or both). In data transmission systems, the communication bandwidth is often the prime resource, and it is with respect to this resource that we view multiaccess protocols here. We give in this paper a unified presentation of the various multiaccess techniques which we group into five categories: 1) fixed assignment techniques, 2) random access techniques, 3) centrally controlled demand assignment techniques, 4) demand assignment techniques with distributed control, and 5) mixed strategies. We discuss their applicability to different enivironments, namely, satellite channels, local area communication networks and multihop store-and-forward broadcast networks, and their applicability to different types of data traffic, namely stream traffic and bursty traffic. We also present the performance of many of the multiaccess protocols in terms of bandwidth utilization and message delay.

Capacity of an IEEE 802.11b wireless LAN supporting VoIP

In this paper we evaluate the capacity of an IEEE 802.11b network carrying voice calls in a wide range of scenarios, including varying delay constraints, channel conditions and voice call quality requirements. We consider both G.711 and G.729 voice encoding schemes and a range of voice packet sizes. We first present an analytical upper bound and, using simulation, show it to be tight in scenarios where channel quality is good and delay constraints are weak or absent. We then use simulation to show that capacity is highly sensitive to the delay budget allocated to packetization and wireless network delays. We also show how channel conditions and voice quality requirements affect the capacity. Selecting the optimum amount of voice data per packet is shown to be a trade-off between throughput and delay constraints: by selecting the packet size appropriately given the delay budget and channel conditions, the capacity can be maximized. Unless a very high voice quality requirement precludes its use, G.729 is shown to allow a capacity greater than or equal to that when G.711 is used, for a given quality requirement.

Expressnet: A High-Performance Integrated-Services Local Area Network

Expressnet is a local area communication network comprising an inbound channel and an outbound channel to which the stations are connected. Stations transmit on the outbound channel and receive on the inbound channel. The inbound channel is connected to the outbound channel so that all signals transmitted on the outbound channel are duplicated on the inbound channel, thus achieving broadcast communication among the stations. In order to transmit on the bus, the stations utilize a distributed access protocol which achieves a conflict-free round-robin scheduling. This protocol is more efficient than existing round-robin Schemes as the time required to switch control from one active user to the next in a round is minimized (on the order of a carrier detection time), and is independent of the end-to-end network propagation delay. This improvement is particularly significant when the channel data rate is so high, or the end-to-end propagation delay is so large, Or the packet size is so small as to render the end-to-end propagation delay a significant fraction of, or larger than, the transmission time of a packet. Moreover, some features of Expressnet make it particularly suitable for voice applications. In view of integrating voice and data, a simple access protocol is described which meets the bandwidth requirement and maximum packet delay constraint for voice communication at all times, while guaranteeing a minimum bandwidth requirement for data traffic. Finally, it is noted that the voice/data access protocol constitutes a highly adaptive allocation scheme of channel bandwidth, which allows data users to recover the bandwidth unused by the voice application. It can be easily extended to accommodate any number of applications, each with its specific requirements.

Assessment of VoIP quality over Internet backbones

As the Internet evolves into a ubiquitous communication infrastructure and provides various services including telephony, it has to stand up to the toll quality standards set by traditional telephone companies. Our objective is to assess to what extent todays Internet meets this expectation. Our assessment is based on delay and loss measurements taken over wide-area backbone networks, considers realistic VoIP scenarios and uses quality measures appropriate for voice. Our findings indicate that although voice services can be adequately provided by some ISPs, a significant number of paths lead to poor performance even for excellent VoIP end-systems. This makes a strong case for special handling of voice traffic on those paths. Even on the good paths, rare loss events can occasionally cause perceptible degradation of voice quality. Finally, the appropriate choice of the playout buffer scheme for each path was found to be of critical importance for the perceived quality.

Assessing the quality of voice communications over Internet backbones

As the Internet evolves into a ubiquitous communication infrastructure and provides various services including telephony, it will be expected to meet the quality standards achieved in the public switched telephone network. Our objective in this paper is to assess to what extent todays Internet meets this expectation. Our assessment is based on delay and loss measurements taken over wide-area backbone networks and uses subjective voice quality measures capturing the various impairments incurred. First, we compile the results of various studies into a single model for assessing the voice-over-IP (VoIP) quality. Then, we identify different types of typical Internet paths and study their VoIP performance. For each type of path, we identify those characteristics that affect the VoIP perceived quality. Such characteristics include the network loss and the delay variability that should be appropriately handled by the playout scheduling at the receiver. Our findings indicate that although voice services can be adequately provided by some ISPs, a significant number of Internet backbone paths lead to poor performance.

Packet Switching in Radio Channels: Part III--Polling and (Dynamic) Split-Channel Reservation Multiple Access

Here we continue the analytic study of packet switching in radio channels which we reported upon m our two previous papers [1], [2] Again we consider a population of terminals communicating with a central station over a packet-switched radio channel. The allocation of bandwidth among the contending terminals can be fixed [e.g., time-division multiple access (TDMA) or frequency-division multiple access (FDMA)], random [e.g., ALOHA or carrier sense multiple access (CSMA)] or centrally controlled (e.g., polling or reservation). In this paper we show that with a large population of bursty users, (as expected) random access is superior to both fixed assignment and polling. We also introduce and analyze a dynamic reservation technique which we call split-channel reservation multiple access (SRMA) which is interesting in that it is both simple and efficient over a large range of system parameters.

Issues in packet radio network design

There are many design choices that must be made in the development of a packet radio network. There is usually no single correct choice, and the decisions are dependent on the environment that the network must work in, the requirements for performance and other functionalities, and the cost and other limitations, in addition, as new hardware and software technologies become available, the parameters governing the decisions change and often result in different selections. This paper outlines a number of design issues and choices available. The intent is to provide an overview of the design decisions that must be made so as to provide a context for the decisions made in a number of existing and developing packet radio networks. It is hoped that this will allow future designs to take advantage of both the wealth of experience available as well as new technologies. Three areas of design decisions are identified. The first area deals with the physical aspects of the network and concentrates on the radio connectivity and channel sharing. The second area deals with the automated management of the network and concentrates on issues such as link management and routing. The third area deals with the interface of the network to the users and some practical aspects of operating and maintaining a network.

Provisioning IP backbone networks to support latency sensitive traffic

To support latency sensitive traffic such as voice, network providers can either use service differentiation to prioritize such traffic or provision their network with enough bandwidth so that all traffic meets the most stringent delay requirements. In the context of wide-area Internet backbones, two factors make overprovisioning an attractive approach. First, the high link speeds and large volumes of traffic make service differentiation complex and potentially costly to deploy. Second, given the degree of aggregation and resulting traffic characteristics, the amount of overprovisioning necessary may not be very large. This study develops a methodology to compute the amount of overprovisioning required to support a given delay requirement. We first develop a model for backbone traffic which is needed to compute the end-to-end delay through the network. The model is validated using 331 one-hour traffic measurements collected from the Sprint IP network. We then develop a procedure which uses this model to find the amount of bandwidth needed on each link in the network so that an end-to-end delay requirement is satisfied. Applying this procedure to the Sprint network, we find that satisfying end-to-end delay requirements as low as 3 ms requires only 15% extra bandwidth above the average data rate of the traffic.