Absfract-Radio communicationis 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.

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Abstract-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 widearea 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.

Abstract-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.

Abstract-As the Internet evolves into a ubiquitous communication infrastructure and provides various services including telephony, it will be expected to stand up to the toll quality standards set by traditional telephone companies. Our objective in this paper is to assess to what extent today's Internet is meeting 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.

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Abstract-We measure and analyze the single-hop packet delay through operational routers in a backbone IP network. First we present our delay measurements through a single router. Then we identify stepby-step the factors contributing to single-hop delay. In addition to packet processing, transmission, and queueing delays, we identify the presence of very large delays due to non-work-conserving router behavior. We use a simple output queue model to separate those delay components. Our step-by-step methodology used to obtain the pure queueing delay is easily applicable to any single-hop delay measurements.After obtaining the queueing delay, we analyze the tail of its distribution, and find that it is long tailed and fits a Weibull distribution with the scale parameter, ¼ , and the shape parameter, ¼ to ¼ . The measured average queueing delay is larger than predicted by M/M/1, M/G/1, and FBM models when the link utilization is below 70%, but its absolute value is quite small.

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 today's 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.Index Terms-Computer networks, Internet, measurements, voice communications, voice over IP (VoIP), voice quality assessment.