Although the Internet was designed to handle non-real time data traffic, it is being used increasingly to carry voice and video. One important class of contributors to this growth are Internet telephones. Critical to more widespread use of Internet telephony is smooth interoperability with the existing telephone network. This interoperability comes through the use of Internet Telephony Gateway’s (ITG’s) which perform protocol translation between an IP network and the Public Switched Telephone Network (PSTN). In order for an IP host to call a user on the PSTN, the IP host must know the IP address of an appropriate gateway. We consider here the problem of finding these gateways. An analysis of a number of protocol architectures is presented, including hierarchical databases, multicast advertisement, routing protocols, and centralized databases. We propose a new protocol architecture, called Brokered Multicast Advertisements (BMA) which serves as a lightweight, scalable mechanism for locating ITG’s. The BMA architecure is general, and can be applied to location of any service across a wide area network. 1

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This paper was written for an Independent Study course. Princy Mehta Overview of Voice over IP Professor Udani February 2001 1 Table of Contents ACRONYMS AND DEFINITIONS............................................................................................................... 3 INTRODUCTION ........................................................................................................................................... 5 IMPLEMENTATION OF VOICE OVER IP............................................................................................... 6 OVERVIEW OF TCP/IP ................................................................................................................................... 6 PACKETIZATION............................................................................................................................................. 7 COMPONENTS OF VO

Abstract—In this paper we analyze packet traces of widely used voice codecs and present analytical source models which describe their output by stochastic processes. Both the G.711 and the G.729.1 codec yield periodic packet streams with a fixed packet size, the G.723.1 as well as the iLBC codec use silence detection leading to an on/off process, and the GSM AMR and the iSAC codec produce periodic packet streams with variable packet sizes. We apply all codecs to a large set of typical speech samples and analyze the output of the codecs statistically. Based on these evaluations we provide quantitative models using standard and modified on/off processes as well as memory Markov chains. Our models are simple and easy to use. They are in good accordance with the original traces as they capture not only the complementary cumulative distribution function (CCDF) of the on/off phase durations and the packet sizes, but also the autocorrelation function (ACF) of consecutive packet sizes as well as the queuing properties of the original traces. In contrast, voice traffic models used in most of today’s simulations or analytical studies fail to reproduce the ACF and the queueing properties of original traces. This possibly leads to underestimation of performance measures like the waiting time or loss probabilities. The models proposed in this paper do not suffer from this shortcoming and present an attractive alternative for use in future performance studies. Index Terms—traffic models, voice codecs, autocorrelation, queuing behavior

A solution to the problem of speech recognition with signals distorted by low-bit rate coders is presented in this paper. A model for the coding-decoding distortion, a HMM compensation method to include this model, and an EM-based adaptation algorithm to estimate this distortion are proposed here. Medium vocabulary continuous-speech speaker-independent recognition experiments with 8 kbps G.729(CS-CELP), 13 kbps RPE-LTP (GSM), 5.3 kbps G723.1, 4.8 kbps FS-1016 and 32 kbps G.726(ADPCM) coders show that the approach described in this paper is able to dramatically reduce the effect of the coding distortion and, in some cases, gives a word accuracy higher than the baseline system with uncoded speech. Finally, the EM estimation algorithm requires only one adapting utterance and the approach described is certainly The evolution and popularity of cellular and TCP/IP networks has created the problem of improving the recognition accuracy for speech distorted by low-bit rate coders. The distortion of coding schemes in speech recognizers is difficult to model and is an open problem that cannot be solved by applying conventional noise cancelling techniques [1] such as spectral subtraction [2], cepstral mean subtraction [3] and RASTA

If highly compressed multimedia streams are transported over packet networks, losses of individual packets can impair the perceptual quality of the received stream in different degrees, depending on the content and context of the lost packet. In this paper, we investigate the impact of individual packet loss on the perceptual speech quality in Voice-over-lP systems using three popular coding types and receiver-side loss concealment algorithms. We set up a testing environment to measure the impairment of individual packet losses and define an appropriate quality metric. We evaluate published algorithms on packet loss quality prediction (DTX, Source-Driven Packet Marking and SPB-DiffMark) and identify their strengths and weaknesses. The quality of a VolP telephone call can be enhanced significant, if a precise packet-loss quality model decides for each VolP packet, how it should be forwarded throughout the network.

This dissertation focuses on how to share a time-division multiplexed wireless channel among packets of different applications with different packet delivery requirements. The novelty of the approach lies in the goal of the allocation: improving user perceived application quality. The approach proposed is based on the paradigm that resource usage that does not improve user perceived quality is ineffective. Utility curves are used to map network service to perceived user quality, expressing the user sensitivity to delivered network service for each application. Based on the history of the network service delivered to a flow, the current perceived quality can be evaluated. Similarly, the quality increase achieved by transmitting a flow’s packet, and the quality decrease caused by the deferral of its transmission for some time, while another flow uses the channel, can be calculated. The packet scheduler proposed—PeLe— allocates the channel to the flow that maximises the sum of the overall quality after transmission of its packet. This accounts for the quality increase of the