@MASTERSTHESIS\{IMM2002-01284, author = "R. D. Ruscior", title = "Voice over {IP} measurements", year = "2002", keywords = "VoIP, Packets, Round Trip Delay, H.323, Jitter, Network, Measurements, Calls", school = "Informatics and Mathematical Modelling, Technical University of Denmark, {DTU}", address = "Richard Petersens Plads, Building 321, {DK-}2800 Kgs. Lyngby", type = "", url = "http://www2.compute.dtu.dk/pubdb/pubs/1284-full.html", abstract = "The Quality of Service is very important for any user, related to Voice over {IP} or any other multimedia application. This thesis presents a solution to develop a tool able to measure this parameter in order to detect upcoming bottleneck problems for a future installation of a VoIP system and to troubleshoot existing VoIP systems. The tool's name is VIPSim. VIPSim is meant to be used to provide extensive information about performance, capacity and problems in an H.323 network. It is also used for alarms, planning and documentation. The thesis is structured in 7 sections: Section 1 presents the general problem of QoS in a VoIP system and shows a possible solution, which is VIPSim. Section 2 presents the architecture of VIPSim. It describes the two components of VIPSim, which are the Sender and the ReÛector. It also shows what VIPSim does and how it does (referring to measurements). It addresses to users of VIPSim in systems troubleshootings. Section 3 goes deeper into VIPSim's implementation, describing its internal design as well as the description of the implementation of some important operations. It addresses to programmers and developers. Section 4 describes some aspects regarding optimizations that were used in {VIP-}Sim's implementation. This section also gives the reader ideas about some algorithms used in VIPSim's implementation. Section 5 shows tests and results obtained from measurements done with {VIP-}Sim. Some of them are meant to determine VIPSim's limitations and requirements. Section 6 contains general conclusions. Section 7 contains a glossary with terms used in this thesis, and some VoIP terminology." }