DOCSLIB.ORG

Application Note Pre-Deployment and Network Readiness Assessment Is

Total Page：16

File Type：pdf, Size：1020Kb

Download full-text PDF Read full-text
Application Note Pre-Deployment and Network Readiness Assessment Is Application Note Contents Title Six Steps To Getting Your Network Overview.......................................................... 1 Ready For Voice Over IP Pre-Deployment and Network Readiness Assessment Is Essential ..................... 1 Date January 2005 Types of VoIP Overview Performance Problems ...... ............................. 1 This Application Note provides enterprise network managers with a six step methodology, including pre- Six Steps To Getting deployment testing and network readiness assessment, Your Network Ready For VoIP ........................ 2 to follow when preparing their network for Voice over Deploy Network In Stages ................................. 7 IP service. Designed to help alleviate or significantly reduce call quality and network performance related Summary .......................................................... 7 problems, this document also includes useful problem descriptions and other information essential for successful VoIP deployment. Pre-Deployment and Network IP Telephony: IP Network Problems, Equipment Readiness Assessment Is Essential Configuration and Signaling Problems and Analog/ TDM Interface Problems. IP Telephony is very different from conventional data applications in that call quality IP Network Problems: is especially sensitive to IP network impairments. Existing network and “traditional” call quality Jitter -- Variation in packet transmission problems become much more obvious with time that leads to packets being the deployment of VoIP service. For network discarded in VoIP end systems; jitter is managers, this means that LANs, access links usually due to network congestion and network equipment may need to be upgraded and that more sophisticated management and Packet Loss -- Packets lost during diagnostic tools are needed when deploying and transmission due to network errors, maintaining VoIP networks. route changes, link failures or random early detection (RED) in routers Types of VoIP Performance Problems Delay -- Overall packet transmission “lag time” that leads to two-way There are three basic categories of conversational difficulty. performance-related problems that can occur in Application Note Six Steps To Getting Your Network Ready For Voice Over IP January 2005 1 Telchemy Application Note Equipment Configuration and For example, a single user accessing a file from a server Signaling Problems: can cause a period of congestion lasting a few seconds. VoIP Endpoint Configuration This, in turn, can cause short-term degradation in --Performance impact of the type of call quality for other users on the network. Given this, CODEC and packet loss concealment it is essential that network managers use performance algorithm, or jitter buffer configuration management tools that can detect and measure these types of network impairments. Router and Firewall Configuration -- Firewalls or incorrectly configured The transient nature of IP problems also routers block VoIP traffic; routers need means that they are not easily reproduced for to be configured to deliver RTP packets analysis once the call is terminated. Unlike in a timely manner traditional POTS, once an end user completes and disconnects an IP call, vital diagnostic information Bandwidth Allocation -- Network about that call and its packet stream is lost. lacks sufficient bandwidth to support Network managers can use packet loss and jitter peak traffic volumes. metrics to determine how bad the call quality was; however, these metrics alone do not provide Analog/TDM Interface Problems: enough diagnostic information to determine why the call was bad. Echo -- “Echo” commonly occurs at the boundary between the digital network (VoIP or TDM) and analog local loops. This becomes very obvious Six Steps To Getting Your Network and annoying with the additional delay Ready For VoIP introduced by the IP network problems previously described. Step 1 Define High-level VoIP Requirements Signal Level -- Abnormally high or low voice signal levels, “clipping,” Your ability to deliver good quality VoIP excessive noise and “echo” occur due performance will depend on patterns of traffic and to incorrectly configured gateway signal usage, existing network capacity, existing data levels. bandwidth and many other factors. The first step is to define what your VoIP deployment will look Network architects and managers should like: address call quality and performance management problems when they plan and deploy their IP What utilization do you expect? networks, but they should be aware that these problems also frequently occur during normal day- Where will gateways be located? to-day network operation. How will internal calls be routed? Many VoIP-related problems are transient in nature and can occur at many places along the network path. How will external calls be routed? 2 Application Note Six Steps To Getting Your Network Ready For Voice Over IP Januaary 2005 Telchemy Application Note What CODECs do you plan to use (e.g. Using low bandwidth CODECs helps reduce G.711, G.729A, iLBC..)? And what bandwidth requirements however due to the bandwidth do these take (including overhead of IP, UDP and RTP protocols, such IP headers - 96kbps for G.711 and reduction is limited. The table below shows typically 24kbps for G.729A)? the effective network bandwidth used by some common CODEC types for different voice frame This should allow you to estimate the amount of sizes. Note that using a longer frame can have network bandwidth required between each location. some impact on delay, hence it is common to use Consider the following example: 10-20mS frames. Scenario: A remote location has 50 Low bit rate CODECs also introduce some users, and no more than 10 users are voice distortion and may be unacceptable for expected to be on the phone at any one extended use. time. A gateway located on the central site will be used for connection to the Step 2 telephone network, and G.711 will be Map Existing Wide Area Network/VPN used. The remote location is connected Capabilities to the central site via IP VPN and has a single T1 to access the IP network. Many call quality-related problems occur in access links or on limited bandwidth WAN or Bandwidth requirements: 10xG.711 VPN links. If significant jitter or delay occurs on streams will require 10 x 96 kbits/s of inter-site connections, this is a strong indicator bandwidth or 960 kbits/s, with some that similar problems will occur during VoIP extra overhead for signaling. If the deployment. Budget bandwidth usage between number of streams increases to 15 then sites and verify that routers can prioritize RTP the bandwidth needed would be 1.44 traffic. Mbits/s (almost the entire T1). What is the bandwidth available between each site? How much data traffic is currently Figure 1: Effective Network carried, (both average levels and peak Bandwidth By CODEC Type For levels)? Different Voice Frame Sizes. CODEC Effective Network Bandwidth (kbits/s) Bandwidth CODEC Type (kbits/s) 5mS Frame 10mS Frame 20mS Frame 30 mS Frame G.711 64 131.2 97.6 80.8 75.1 G.729A 8 - 41.6 24.8 19.1 G.723.1 6.3 - - - 17.4 iLBC 15.2 - - 32 - Application Note Six Steps To Getting Your Network Ready For Voice Over IP January 2005 3 Telchemy Application Note Is sufficient bandwidth available to Examine Ethernet switch statistics for support both the current data traffic and evidence of packet errors or excessive the estimated bandwidth needed for collisions and upgrade equipment voice? accordingly. Is there any capacity for growth or for If Wireless LANs are being used, be aware that unexpected peaks in activity? these can introduce significant impairments into the packet stream and should be checked carefully Will the addition of voice traffic have before using them for Voice traffic. an adverse effect on data application performance, i.e. if there is significantly Step 4 less bandwidth for data applications, Verify Inter-Site Readiness then how will they be affected? Pre-deployment testing is essential; however, you Map VoIP bandwidth requirements onto need to perform Steps 1 to 3 first. Many problems available bandwidth. are predictable by simply verifying that sufficient bandwidth is available to carry the expected traffic 10xG.711 streams will require 10 x 96 kbits/s level. of bandwidth or 960 kbits/s, with some extra overhead for signaling. If the number of streams When conducting a pre-deployment test, it is increases to 15, then the bandwidth needed important that the test: would be 1.44 Mbits/s (almost the entire T1). This will leave approximately 500 kbits/s for Is conducted over a sufficient period data traffic. If more than the expected 10 users of time to assess network readiness are simultaneously on the phone, then problems under a variety of “typical” network may start to occur with data applications due to conditions. Conditions vary through the insufficient bandwidth. And if more than 15 users day and through the week, and network are on the phone, then voice quality will degrade problems are often transient in nature. very quickly. There is no substitute for a sufficient amount of testing time. Step 3 Verify LAN readiness Is conducted under network load conditions that are similar to those Use a switched 100BaseT Ethernet expected post-deployment. There is architecture, potentially using VLAN to separate little point in testing a single VoIP call voice and data. Even with the use of switched when you expect that there will be 20 Ethernet, problems can still occur due to duplex calls. mismatch, excessively long Ethernet
Load more

Recommended publications
  • Latency and Throughput Optimization in Modern Networks: a Comprehensive Survey Amir Mirzaeinnia, Mehdi Mirzaeinia, and Abdelmounaam Rezgui
    READY TO SUBMIT TO IEEE COMMUNICATIONS SURVEYS & TUTORIALS JOURNAL 1 Latency and Throughput Optimization in Modern Networks: A Comprehensive Survey Amir Mirzaeinnia, Mehdi Mirzaeinia, and Abdelmounaam Rezgui Abstract—Modern applications are highly sensitive to com- On one hand every user likes to send and receive their munication delays and throughput. This paper surveys major data as quickly as possible. On the other hand the network attempts on reducing latency and increasing the throughput. infrastructure that connects users has limited capacities and These methods are surveyed on different networks and surrond- ings such as wired networks, wireless networks, application layer these are usually shared among users. There are some tech- transport control, Remote Direct Memory Access, and machine nologies that dedicate their resources to some users but they learning based transport control, are not very much commonly used. The reason is that although Index Terms—Rate and Congestion Control , Internet, Data dedicated resources are more secure they are more expensive Center, 5G, Cellular Networks, Remote Direct Memory Access, to implement. Sharing a physical channel among multiple Named Data Network, Machine Learning transmitters needs a technique to control their rate in proper time. The very first congestion network collapse was observed and reported by Van Jacobson in 1986. This caused about a I. INTRODUCTION thousand time rate reduction from 32kbps to 40bps [3] which Recent applications such as Virtual Reality (VR), au- is about a thousand times rate reduction. Since then very tonomous cars or aerial vehicles, and telehealth need high different variations of the Transport Control Protocol (TCP) throughput and low latency communication.
    [Show full text]
  • Lab 6: Understanding Traditional TCP Congestion Control (HTCP, Cubic, Reno)
    NETWORK TOOLS AND PROTOCOLS Lab 6: Understanding Traditional TCP Congestion Control (HTCP, Cubic, Reno) Document Version: 06-14-2019 Award 1829698 “CyberTraining CIP: Cyberinfrastructure Expertise on High-throughput Networks for Big Science Data Transfers” Lab 6: Understanding Traditional TCP Congestion Control Contents Overview ............................................................................................................................. 3 Objectives............................................................................................................................ 3 Lab settings ......................................................................................................................... 3 Lab roadmap ....................................................................................................................... 3 1 Introduction to TCP ..................................................................................................... 3 1.1 TCP review ............................................................................................................ 4 1.2 TCP throughput .................................................................................................... 4 1.3 TCP packet loss event ........................................................................................... 5 1.4 Impact of packet loss in high-latency networks ................................................... 6 2 Lab topology...............................................................................................................
    [Show full text]
  • Adaptive Method for Packet Loss Types in Iot: an Naive Bayes Distinguisher
    electronics Article Adaptive Method for Packet Loss Types in IoT: An Naive Bayes Distinguisher Yating Chen , Lingyun Lu *, Xiaohan Yu * and Xiang Li School of Computer and Information Technology, Beijing Jiaotong University, Beijing 100044, China; [email protected] (Y.C.); [email protected] (X.L.) * Correspondence: [email protected] (L.L.); [email protected] (X.Y.) Received: 31 December 2018; Accepted: 23 January 2019; Published: 28 January 2019 Abstract: With the rapid development of IoT (Internet of Things), massive data is delivered through trillions of interconnected smart devices. The heterogeneous networks trigger frequently the congestion and influence indirectly the application of IoT. The traditional TCP will highly possible to be reformed supporting the IoT. In this paper, we find the different characteristics of packet loss in hybrid wireless and wired channels, and develop a novel congestion control called NB-TCP (Naive Bayesian) in IoT. NB-TCP constructs a Naive Bayesian distinguisher model, which can capture the packet loss state and effectively classify the packet loss types from the wireless or the wired. More importantly, it cannot cause too much load on the network, but has fast classification speed, high accuracy and stability. Simulation results using NS2 show that NB-TCP achieves up to 0.95 classification accuracy and achieves good throughput, fairness and friendliness in the hybrid network. Keywords: Internet of Things; wired/wireless hybrid network; TCP; naive bayesian model 1. Introduction The Internet of Things (IoT) is a new network industry based on Internet, mobile communication networks and other technologies, which has wide applications in industrial production, intelligent transportation, environmental monitoring and smart homes.
    [Show full text]
  • Congestion Control Overview
    ELEC3030 (EL336) Computer Networks S Chen Congestion Control Overview • Problem: When too many packets are transmitted through a network, congestion occurs At very high traffic, performance collapses Perfect completely, and almost no packets are delivered Maximum carrying capacity of subnet • Causes: bursty nature of traffic is the root Desirable cause → When part of the network no longer can cope a sudden increase of traffic, congestion Congested builds upon. Other factors, such as lack of Packets delivered bandwidth, ill-configuration and slow routers can also bring up congestion Packets sent • Solution: congestion control, and two basic approaches – Open-loop: try to prevent congestion occurring by good design – Closed-loop: monitor the system to detect congestion, pass this information to where action can be taken, and adjust system operation to correct the problem (detect, feedback and correct) • Differences between congestion control and flow control: – Congestion control try to make sure subnet can carry offered traffic, a global issue involving all the hosts and routers. It can be open-loop based or involving feedback – Flow control is related to point-to-point traffic between given sender and receiver, it always involves direct feedback from receiver to sender 119 ELEC3030 (EL336) Computer Networks S Chen Open-Loop Congestion Control • Prevention: Different policies at various layers can affect congestion, and these are summarised in the table • e.g. retransmission policy at data link layer Transport Retransmission policy • Out-of-order caching
    [Show full text]
  • The Effects of Different Congestion Control Algorithms Over Multipath Fast Ethernet Ipv4/Ipv6 Environments
    Proceedings of the 11th International Conference on Applied Informatics Eger, Hungary, January 29–31, 2020, published at http://ceur-ws.org The Effects of Different Congestion Control Algorithms over Multipath Fast Ethernet IPv4/IPv6 Environments Szabolcs Szilágyi, Imre Bordán Faculty of Informatics, University of Debrecen, Hungary [email protected] [email protected] Abstract The TCP has been in use since the seventies and has later become the predominant protocol of the internet for reliable data transfer. Numerous TCP versions has seen the light of day (e.g. TCP Cubic, Highspeed, Illinois, Reno, Scalable, Vegas, Veno, etc.), which in effect differ from each other in the algorithms used for detecting network congestion. On the other hand, the development of multipath communication tech- nologies is one today’s relevant research fields. What better proof of this, than that of the MPTCP (Multipath TCP) being integrated into multiple operating systems shortly after its standardization. The MPTCP proves to be very effective for multipath TCP-based data transfer; however, its main drawback is the lack of support for multipath com- munication over UDP, which can be important when forwarding multimedia traffic. The MPT-GRE software developed at the Faculty of Informatics, University of Debrecen, supports operation over both transfer protocols. In this paper, we examine the effects of different TCP congestion control mechanisms on the MPTCP and the MPT-GRE multipath communication technologies. Keywords: congestion control, multipath communication, MPTCP, MPT- GRE, transport protocols. 1. Introduction In 1974, the TCP was defined in RFC 675 under the Transmission Control Pro- gram name. Later, the Transmission Control Program was split into two modular Copyright © 2020 for this paper by its authors.
    [Show full text]
  • What Causes Packet Loss IP Networks
    What Causes Packet Loss IP Networks (Rev 1.1) Computer Modules, Inc. DVEO Division 11409 West Bernardo Court San Diego, CA 92127, USA Telephone: +1 858 613 1818 Fax: +1 858 613 1815 www.dveo.com Copyright © 2016 Computer Modules, Inc. All Rights Reserved. DVEO, DOZERbox, DOZER Racks and DOZER ARQ are trademarks of Computer Modules, Inc. Specifications and product availability are subject to change without notice. Packet Loss and Reasons Introduction This Document To stream high quality video, i.e. to transmit real-time video streams, over IP networks is a demanding endeavor and, depending on network conditions, may result in packets being dropped or “lost” for a variety of reasons, thus negatively impacting the quality of user experience (QoE). This document looks at typical reasons and causes for what is generally referred to as “packet loss” across various types of IP networks, whether of the managed and conditioned type with a defined Quality of Service (QoS), or the “unmanaged” kind such as the vast variety of individual and interconnected networks across the globe that collectively constitute the public Internet (“the Internet”). The purpose of this document is to encourage operators and enterprises that wish to overcome streaming video quality problems to explore potential solutions. Proven technology exists that enables transmission of real-time video error-free over all types of IP networks, and to perform live broadcasting of studio quality content over the “unmanaged” Internet! Core Protocols of the Internet Protocol Suite The Internet Protocol (IP) is the foundation on which the Internet was built and, by extension, the World Wide Web, by enabling global internetworking.
    [Show full text]
  • Packet Loss Burstiness: Measurements and Implications for Distributed Applications
    Packet Loss Burstiness: Measurements and Implications for Distributed Applications David X. Wei1,PeiCao2,StevenH.Low1 1Division of Engineering and Applied Science 2Department of Computer Science California Institute of Technology Stanford University Pasadena, CA 91125 USA Stanford, CA 94305 USA {weixl,slow}@cs.caltech.edu [email protected] Abstract sources. The most commonly used control protocol for re- liable data transmission is Transmission Control Protocol Many modern massively distributed systems deploy thou- (TCP) [18], with a variety of congestion control algorithms sands of nodes to cooperate on a computation task. Net- (Reno [5], NewReno [11], etc.), and implementations (e.g. work congestions occur in these systems. Most applications TCP Pacing [16, 14]). The most commonly used control rely on congestion control protocols such as TCP to protect protocol for unreliable data transmission is TCP Friendly the systems from congestion collapse. Most TCP conges- Rate Control (TFRC) [10]. These algorithms all use packet tion control algorithms use packet loss as signal to detect loss as the congestion signal. In designs of these protocols, congestion. it is assumed that the packet loss process provides signals In this paper, we study the packet loss process in sub- that can be detected by every flow sharing the network and round-trip-time (sub-RTT) timescale and its impact on the hence the protocols can achieve fair share of the network loss-based congestion control algorithms. Our study sug- resource and, at the same time, avoid congestion collapse. gests that the packet loss in sub-RTT timescale is very bursty. This burstiness leads to two effects.
    [Show full text]
  • A Network Congestion Control Protocol (NCP)
    A Network Congestion control Protocol (NCP) Debessay Fesehaye, Klara Nahrstedt and Matthew Caesar Department of Computer Science UIUC, 201 N Goodwin Ave Urbana, IL 61801-2302, USA Email:{dkassa2,klara,caesar}@cs.uiuc.edu Abstract The transmission control protocol (TCP) which is the dominant congestion control protocol at the transport layer is proved to have many performance problems with the growth of the Internet. TCP for instance results in throughput degradation for high bandwidth delay product networks and is unfair for flows with high round trip delays. There have been many patches and modifications to TCP all of which inherit the problems of TCP in spite of some performance improve- ments. On the other hand there are clean-slate design approaches of the Internet. The eXplicit Congestion control Protocol (XCP) and the Rate Control Protocol (RCP) are the prominent clean slate congestion control protocols. Nonetheless, the XCP protocol is also proved to have its own performance problems some of which are its unfairness to long flows (flows with high round trip delay), and many per-packet computations at the router. As shown in this paper RCP also makes gross approximation to its important component that it may only give the performance reports shown in the literature for specific choices of its parameter values and traffic patterns. In this paper we present a new congestion control protocol called Network congestion Control Protocol (NCP). We show that NCP can outperform both TCP, XCP and RCP in terms of among other things fairness and file download times. Keywords: Congestion Control, clean-slate, fairness, download times.
    [Show full text]
  • How to Properly Measure and Correct Packet Loss
    White Paper How to Properly Measure and Correct Packet Loss Silver Peak avoids common packet delivery issues across the WAN White Paper How to Properly Measure and Correct Packet Loss Silver Peak avoids common packet delivery issues across the WAN Shared Wide Area Networks (WANs), like MPLS and Internet VPNs, are prone to network congestion during periods of heavy utilization. When different traffic is vying for limited shared resources, packets inevitably will be dropped or delivered out of order, a concept known as “packet loss”. As more enterprises deploy real-time traffic, like voice, video and other unified communications, packet loss is an increasingly significant problem. That is because dropped and out-of-order packets degrade the quality of latency sensitive applications – e.g. packet loss causes video pixilation, choppy phone calls, or terminated sessions. In addition, loss is increasingly problematic as more enterprises look to lower disaster recovery costs by performing data replication over shared WANs. As figure 1 shows, when a small amount of packet loss (0.1%) is combined with a marginal amount of latency (50 ms), data throughput will never exceed 10 Mbps per flow – regardless of how much bandwidth is available. This can severely hamper key enterprise initiatives that require high sustained data throughput, like replication. The problem is that most enterprises don’t know packet loss exists on their WANs as they don’t have the proper tools for measuring it. (Service Providers only exacerbate the problem by offering SLAs that are based on monthly averages, which can be quite misleading. An enterprise can experience several hours of loss during several work days and still statistically have around 0% loss over the course of a month).
    [Show full text]
  • The Need for Realism When Simulating Network Congestion
    The Need for Realism when Simulating Network Congestion Kevin Mills Chris Dabrowski NIST NIST Gaithersburg, MD 20899 Gaithersburg, MD 20899 [email protected] [email protected] throughout the network. The researchers identify various ABSTRACT Many researchers use abstract models to simulate network signals that arise around the critical point. Such signals congestion, finding patterns that might foreshadow onset of could foreshadow onset of widespread congestion. These congestion collapse. We investigate whether such abstract developments appear promising as a theoretical basis for monitoring methods that could be deployed to warn of models yield congestion behaviors sufficiently similar to impending congestion collapse. Despite showing promise, more realistic models. Beginning with an abstract model, we add elements of realism in various combinations, questions surround this research, as the models are quite culminating with a high-fidelity simulation. By comparing abstract, bearing little resemblance to communication congestion patterns among combinations, we illustrate networks deployed based on modern technology. We congestion spread in abstract models differs from that in explore these questions by examining the influence of realistic models. We identify critical elements of realism realism on congestion spread in network simulations. needed when simulating congestion. We demonstrate a We begin with an abstract network simulation from the means to compare congestion patterns among simulations literature. We add realism elements in combinations, covering diverse configurations. We hope our contributions culminating with a high-fidelity simulation, also from the lead to better understanding of the need for realism when literature. By comparing patterns of congestion among the simulating network congestion. combinations, we explore a number of questions.
    [Show full text]
  • TCP Congestion Control Scheme for Wireless Networks Based on TCP Reserved Field and SNR Ratio
    International Journal of Research and Reviews in Information Sciences (IJRRIS), ISSN: 2046-6439, Vol. 2, No. 2, June 2012 http://www.sciacademypublisher.com/journals/index.php/IJRRIS/article/download/817/728 TCP Congestion Control Scheme for Wireless Networks based on TCP Reserved Field and SNR Ratio Youssef Bassil LACSC – Lebanese Association for Computational Sciences, Registered under No. 957, 2011, Beirut, Lebanon Email: [email protected] Abstract – Currently, TCP is the most popular and widely used network transmission protocol. In actual fact, about 90% of connections on the internet use TCP to communicate. Through several upgrades and improvements, TCP became well optimized for the very reliable wired networks. As a result, TCP considers all packet timeouts in wired networks as due to network congestion and not to bit errors. However, with networking becoming more heterogeneous, providing wired as well as wireless topologies, TCP suffers from performance degradation over error-prone wireless links as it has no mechanism to differentiate error losses from congestion losses. It therefore considers all packet losses as due to congestion and consequently reduces the burst of packet, diminishing at the same time the network throughput. This paper proposes a new TCP congestion control scheme appropriate for wireless as well as wired networks and is capable of distinguishing congestion losses from error losses. The proposed scheme is based on using the reserved field of the TCP header to indicate whether the established connection is over a wired or a wireless link. Additionally, the proposed scheme leverages the SNR ratio to detect the reliability of the link and decide whether to reduce packet burst or retransmit a timed-out packet.
    [Show full text]
  • Simulation and Analysis of Loss in Ip Networks
    SIMULATION AND ANALYSIS OF LOSS IN IP NETWORKS by Velibor Markovski B.Sc., University \Sts. Cyril and Methodius", Macedonia, 1996 a thesis submitted in partial fulfillment of the requirements for the degree of Master of Applied Science in the School of Engineering Science c Velibor Markovski 2000 SIMON FRASER UNIVERSITY October 2000 All rights reserved. This work may not be reproduced in whole or in part, by photocopy or other means, without the permission of the author. APPROVAL Name: Velibor Markovski Degree: Master of Applied Science Title of thesis: Simulation and analysis of loss in IP networks Examining Committee: Prof. Mehrdad Saif, Committee-Chair, Chairman Prof. Ljiljana Trajkovi´c, Senior Supervisor Prof. Stephen Hardy, Supervisor School of Engineering Science Prof. Joseph G. Peters, Supervisor School of Computing Science Prof. Tsunehiko (Tiko) Kameda, External exam- iner School of Computing Science Date Approved: ii Abstract The primary goal of this research is to simulate and analyze the loss process in Internet Protocol (IP) networks. Obtaining insight into the loss patterns in a network is important for measuring the network performance during transfers of multimedia traffic. In order to analyze the packet loss process, we collected specific information about when loss appears in the network, which packets are lost, whether the loss is clustered in bursts (episodes), how long these loss episodes are, and what the impact of network protocols on loss is. Of particular interest were consecutive packet losses because they contribute to the prolonged loss episodes, which adversely affect multimedia traffic. We expect that having better knowledge about the loss patterns may help design decisions for enabling or improving Quality of Service (QoS) support in packet networks.
    [Show full text]