Assessing readiness of IP networks to support desktop videoconferencing using OPNET

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Abstract

OPNET is a powerful network design and simulation tool that has gained popularity in industry and academia. However, there exists no known simulation approach on how to deploy a popular real-time network service such as videoconferencing. This paper demonstrates how OPNET can be leveraged to assess the readiness of existing IP networks to support desktop videoconference. To date, OPNET does not have built-in features to support videoconferencing or its deployment. The paper offers remarkable details on how to model and configure OPNET for such a purpose. The paper considers two types of video traffic (viz. fixed and empirical video packet sizes). Empirical video packet sizes are collected from well-known Internet traffic traces. The paper presents in-depth analysis and interpretation of simulation results and shows how to draw proper engineering conclusions.

Introduction

The deployment of videoconferencing over IP network in both industry and academia has been increasing rapidly. Desktop videoconferencing applications range from internal company communications, educating and training remote employees, to telecommuting. It can eliminate certain travel requirements, thereby cutting costs. Desktop videoconferencing takes advantage of a key workplace tool that is the PC. In the past few years, an H.323 standard was introduced by the ITU, and thus paved the way to the fast growth and deployment of videoconferencing. H.323 is a full suite of protocols developed by ITU to define how real-time multimedia communications, such as videoconferencing, can be exchanged over packet-switched networks (Recommendation H.323, 1998).

It is very advantageous and cost effective to deploy desktop videoconferencing over their existing IP networks. It is easier to run, manage, and maintain. However, one has to keep in mind that IP networks are best-effort networks that were designed for non-real time applications. On the other hand, videoconferencing requires timely packet delivery with low latency, jitter, packet loss, and sufficient bandwidth. To achieve this goal, an efficient deployment of videoconferencing must ensure these real-time traffic requirements can be guaranteed over new or existing IP networks.

Videoconferencing places a high demand on network resources. When deploying such a network service, network architects, managers, planners, designers, and engineers are faced with common strategic, and sometimes challenging, questions. What are the QoS requirements for videoconferencing? How will the new videoconferencing load impact the QoS for currently running network services and applications? Will my existing network support videoconferencing and satisfy the standardized QoS requirements? If so, how many videoconferencing sessions can the network support before upgrading prematurely any part of the existing network hardware?

A number of commercial tools have been developed to address issues related to videoconferencing deployment over data networks. We summarize most popular commercial tools. EURESOM Jupitor II (Eurescom H.323 Studies) has a provision to test end-to-end Quality of Service (QoS) for Network-QoS-aware applications over IP networks. It considers the relationship between users’ perception and network performance. NetIQ's Vivinet Assessor (NetIQ) generates RTP streams to mimic VoIP traffic between pairs of hosts and assesses the quality of these synthetic calls. BMC PATROL DashBoard (BMCsoftware) analyzes the impact of multimedia services on the existing network. This tool can quickly identify specific problems on the network that impact application performance. Spirent's IPTV (Spirent Communications Resources) system is a product that includes various features like video infrastructure testing, IPTV video quality testing, firewall and video server load testing. RADVISION (Radvision H.323 Protocol Toolkit) offers tightly integrated infrastructure processing components called viaIP, for desktop and meeting room conferencing. Also other companies that provide VVoIP testing are Omegon, Lucent VitalSuite (Lucent Technologies), and ViDeNet (VideNet Scout Resources). “H.323 Beacon” tool (Calyam et al., 2004) is a open-source tool for assessing performance of desktop videoconferencing sessions using H.323 traffic emulation.

For the most part, these tools use two common approaches in assessing the deployment of videoconferencing into the existing network. One approach is based on first performing network measurements and then predicting the network readiness for supporting videoconferencing. The prediction of the network readiness is based on assessing the health of network elements. The second approach is based on injecting real videoconferencing traffic into existing network and measuring the resulting delay, jitter, and loss.

Other than the cost associated with the commercial tools, none of the commercial tools offers a comprehensive approach for successful VoIP deployment. In particular, none gives any prediction for the total number of calls that can be supported by the network taking into account important design and engineering factors. These factors include VoIP flow and call distribution, future growth capacity, performance thresholds, and impact background traffic. This paper attempts to address those important factors utilizing OPNET simulation.

In this paper, we demonstrate how the popular OPNET simulation tool can be leveraged to assess the readiness of existing data networks to support videoconferencing. To the best of authors’ knowledge and to date, OPNET modeler does not have built-in features to support videoconferencing or deployment of real-time services. In the literature, there exists no known simulation approach on how to deploy a popular real-time network service such as videoconferencing. OPNET has gained considerable popularity in academia as it is being offered free of charge to academic institutions. That has given OPNET an edge over DES NS2 in both market place and academia. Another reason to choose OPNET is the fact that OPNET contains a vast amount of models of commercially available network elements, and has various real-life network configuration capabilities. This makes the simulation of real-life network environment close to reality. Other features of OPNET include GUI interface, comprehensive library of network protocols and models, source code for all models, graphical results and statistics, etc.

In previously related work Salah (2006), an analytic approach based on the principles of queuing networks was presented to determine approximately the number of video sessions an existing data network can support. In sharp contrast to Salah (2006), this paper primarily focuses on showing how to deploy successfully videoconferencing using OPNET modeling and simulation. This paper considers two types of traffic (viz. traffic of fixed video packet sizes as that used for the analytic approach describe in Salah (2006) and also traffic of variable video packet sizes measured from well-known traffic traces). This paper discusses in great detail the simulation configuration, setup, and generation of traffic for videoconferencing. Such information can be extremely useful for network researchers and engineers who are interested in deploying videoconferencing. The paper also gives in-depth analysis and interpretations of OPNET simulation results.

Fig. 1 illustrates a typical network infrastructure of a small- to medium-sized company residing in a high-rise building with the minimal added videoconferencing components of a H.323 gatekeeper and H.323 workstations (Recommendation H.323, 1998; Abler and Wells, 1999; Cisco Systems). The gatekeeper node handles signaling for establishing, terminating, and authorizing connections of video sessions, as well as imposing maximum bandwidth for each session. H.323 workstations or multimedia PCs have H.323 voice and video software and are equipped with a camera and a microphone. The network is Ethernet-based and has Layer-2 Ethernet switches connected by a router. The router is Cisco 2621, and the switches are 3Com Superstack 3300. All the links are switched Ethernet 100 Mbps full duplex. Shared links are never suitable for real-time applications. The network shown is realistic and used as a case study only; however, our work presented in this paper can be adopted easily for larger and general networks by following the same principles, guidelines, and concepts laid out in this paper.

An important step that plays a factor in determining the number of sessions to be supported is the flow of sessions (or calls) and their distribution. Traffic flow has to do with the path that session travels through. Session distribution has to do with the percentage of sessions to be established within and outside of a floor, building, or department. For our example, we will assume that the generation of sessions is symmetric for all three floors. The intra-floor traffic will constitute 20% of overall traffic, and the other 80% will constitute inter-floor traffic. Such a distribution can be described in a simple probability tree shown in Fig. 2.

Throughout our work, we assume voice and video calls are symmetric. We also ignore the signaling traffic generated by the gatekeeper. We consider the worst-case scenario for videoconferencing traffic. The signaling traffic involving the gatekeeper is only generated prior to the establishment of the session and when the session is finished. This traffic is relatively limited and small compared to the actual voice call traffic. In general, the gatekeeper generates no signaling traffic throughout the duration of the videoconferencing session for an already established on-going session (Goode, 2002). In order to allow for future growth, we will consider a 25% growth factor for all network elements including router, switches, and links. This factor will be taken into account in our simulation study.

The rest of the paper is organized as follows. Section 2 describes important key design issues and requirements for videoconferencing that play a role in assessing the network readiness. These are primarily the bandwidth and delay bounds. Section 3 is the detailed simulation work of OPNET. The section describes in detail the simulation model, configuration, setup, and generation of traffic for videoconferencing. The section also interprets simulation results for both fixed and empirical video packet sizes. Section 4 concludes the study and identifies future work.

Section snippets

Delay and bandwidth requirements

For deploying a new network service such as desktop videoconferencing, one has to characterize first two important metrics. First is the available bandwidth. Second is the end-to-end delay. The actual number of videoconferencing sessions that the network can sustain and support is bounded by those two metrics. Depending on the network under study, either the available bandwidth or delay can be the key dominant factor in determining the number of sessions that can be supported. In this paper, we

Simulation study

For our simulation study, we use MIL3's OPNET Modeler simulation package,1 Release 8.0.C OPNET Technologies. This section describes in detail simulation model, traffic model, various simulation configurations, as well as the simulation results.

Conclusion

The paper described in great detail how OPNET can be utilized to assess the readiness of existing IP networks to support desktop videoconferencing. The paper offered extensive interpretations and analysis of simulation results and showed how to draw proper conclusions. Two scenarios of traffic were used for OPNET simulation (viz. fixed and empirical video packet sizes). Videoconferencing traffic of fixed size of 1344 bytes for video packets gave similar outcome (in terms of number of

Acknowledgments

This work has been supported and funded by King Fahd University of Petroleum & Minerals under Project #ICS/VOICE/304.

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