Comparing the Efficiency of IP and ATM Telephony

Circuit switching, suited to providing real-time services due to the low and fixed switching delay, is not cost effective for building integrated services networks bursty data traffic because it is based on static allocation of resources which is not efficient with bursty data traffic. Moreover, since current circuit switching technologies handle flows at rates which are integer multiples of 64 kb/s, low bit rate voice encoding cannot be taken advantage of without aggregating multiple phone calls on a single channel. This work explores the real-time efficiency of IP telephony, i.e. the volume of voice traffic with deterministically guaranteed quality related to the amount of network resources used. IP and ATM are taken into consideration as packet switching technology for carrying compressed voice and it is compared to circuit switching carrying PCM (64 Kb/s) encoded voice. ADPCM32 is the voice encoding scheme used throughout most of the paper. The impact of several network parameters, among which the number of hops traversed by a call, on the real-time efficiency is studie

Survey of traffic control schemes and error control schemes for ATM networks

Among the techniques proposed for B-ISDN transfer mode, ATM concept is considered to be the most promising transfer technique because of its flexibility and efficiency. This paper surveys and reviews a number of topics related to ATM networks. Those topics cover congestion control, provision of multiple classes of traffic, and error control. Due to the nature of ATM networks, those issues are far more challenging than in conventional networks. Sorne of the more promising solutions to those issues are surveyed, and the corresponding results on performance are summarized. Future research problems in ATM protocol aspect are also presented

Survey of congestion control techniques for an ATM network

The emerging broadband integrated services digital network is expected to adopt ATM (Asynchronous Transfer Mode) as the transport network. This new network must support several classes of service with varying delay and loss requirements. It must also operate with link speeds in the hundreds of megabits per second and be scalable up to potential link speeds on the order of gigabits per second. The requirements to support multiple services and high speed make the congestion control in an ATM network difficult. This paper reviews sorne of the techniques for prevention and control of congestion in an ATM network

Elastic calls in an integrated services network: the greater the call size variability the better the QoS

We study a telecommunications network integrating prioritized stream calls and delay tolerant elastic calls that are served with the remaining (varying) service capacity according to a processor sharing discipline. The remarkable observation is presented and analytically supported that the expected elastic call holding time is decreasing in the variability of the elastic call size distribution. As a consequence, network planning guidelines or admission control schemes that are developed based on deterministic or lightly variable elastic call sizes are likely to be conservative and inefficient, given the commonly acknowledged property of e.g.\ \textsc{www}\ documents to be heavy tailed. Application areas of the model and results include fixed \textsc{ip} or \textsc{atm} networks and mobile cellular \textsc{gsm}/\textsc{gprs} and \textsc{umts} networks. \u

On-board congestion control for satellite packet switching networks

It is desirable to incorporate packet switching capability on-board for future communication satellites. Because of the statistical nature of packet communication, incoming traffic fluctuates and may cause congestion. Thus, it is necessary to incorporate a congestion control mechanism as part of the on-board processing to smooth and regulate the bursty traffic. Although there are extensive studies on congestion control for both baseband and broadband terrestrial networks, these schemes are not feasible for space based switching networks because of the unique characteristics of satellite link. Here, we propose a new congestion control method for on-board satellite packet switching. This scheme takes into consideration the long propagation delay in satellite link and takes advantage of the the satellite's broadcasting capability. It divides the control between the ground terminals and satellite, but distributes the primary responsibility to ground terminals and only requires minimal hardware resource on-board satellite

Performance of voice and video conferencing over ATM and gigabit ethernet backbone networks

Gigabit Ethernet and ATM network technologies have been modeled as campus network backbones for the simulation-based comparison of their performance. Real-time voice and video conferencing traffic is used to compare the performance of both backbone technologies in terms of response times and packet end-to-end delays. Simulation results show that Gigabit Ethernet has been able to perform the same and in some cases better than ATM as a backbone network for video and voice conferencing providing network designers with a cheaper solution to meet the growing needs of bandwidth-hungry applications in a campus environment

QoS Based Capacity Enhancement for WCDMA Network with Coding Scheme

The wide-band code division multiple access (WCDMA) based 3G and beyond cellular mobile wireless networks are expected to provide a diverse range of multimedia services to mobile users with guaranteed quality of service (QoS). To serve diverse quality of service requirements of these networks it necessitates new radio resource management strategies for effective utilization of network resources with coding schemes. Call admission control (CAC) is a significant component in wireless networks to guarantee quality of service requirements and also to enhance the network resilience. In this paper capacity enhancement for WCDMA network with convolutional coding scheme is discussed and compared with block code and without coding scheme to achieve a better balance between resource utilization and quality of service provisioning. The model of this network is valid for the real-time (RT) and non-real-time (NRT) services having different data rate. Simulation results demonstrate the effectiveness of the network using convolutional code in terms of capacity enhancement and QoS of the voice and video services.Comment: 10 Pages, VLSICS Journa

DTMsim - DTM channel simulation in ns

Dynamic Transfer Mode (DTM) is a ring based MAN technology that provides a channel abstraction with a dynamically adjustable capacity. TCP is a reliable end to end transport protocol capable of adjusting its rate. The primary goal of this work is investigate the coupling of dynamically allocating bandwidth to TCP flows with the affect this has on the congestion control mechanism of TCP. In particular we wanted to find scenerios where this scheme does not work, where either all the link capacity is allocated to TCP or congestion collapse occurs and no capacity is allocated to TCP. We have created a simulation environment using ns-2 to investigate TCP over networks which have a variable capacity link. We begin with a single TCP Tahoe flow over a fixed bandwidth link and progressively add more complexity to understand the behaviour of dynamically adjusting link capacity to TCP and vice versa

A connection-level call admission control using genetic algorithm for MultiClass multimedia services in wireless networks

Call admission control in a wireless cell in a personal communication system (PCS) can be modeled as an M/M/C/C queuing system with m classes of users. Semi-Markov Decision Process (SMDP) can be used to optimize channel utilization with upper bounds on handoff blocking probabilities as Quality of Service constraints. However, this method is too time-consuming and therefore it fails when state space and action space are large. In this paper, we apply a genetic algorithm approach to address the situation when the SMDP approach fails. We code call admission control decisions as binary strings, where a value of “1” in the position i (i=1,…m) of a decision string stands for the decision of accepting a call in class-i; a value of “0” in the position i of the decision string stands for the decision of rejecting a call in class-i. The coded binary strings are feed into the genetic algorithm, and the resulting binary strings are founded to be near optimal call admission control decisions. Simulation results from the genetic algorithm are compared with the optimal solutions obtained from linear programming for the SMDP approach. The results reveal that the genetic algorithm approximates the optimal approach very well with less complexity