Lab 3 TCOM631 Overview The goal of this lab is to teach you how simulation and modeling can be used in analyzing applications and networks. You will use Riverbed (OPNET) Modeler Academic Edition, simulation and modeling software, to study performance of several voice codes given different IP network conditions (packet loss, network latency). MOS is a typical measure of this performance. You will also make some changes to SIP and H.323 models and observe their operation. All of these modeling scenarios will use something know as Discrete Event Simulation (DES). Objectives Get familiar with the software and VoIP modeling capabilities Analyze performance of the following voice codecs: G.711 Silence, G.726 32K, G.726 32K Silence, G.728 16K, G.723.1 5.3K, G.729A, G.729A Silence Specify operation of SIP and H323 signaling/control protocols and measure GoS and signaling latency Discuss additional details, answer several questions on the way through the lab, collect results and produce reports with your observations Write your names here: Page 1 of 17
Instructions 1) Open Riverbed Modeler Academic Edition 17.5 and accept software agreement. 2) Go to File Open and open project named VoIP_design_TCOM590 (located in C:\App_Data\Cadence\SPB_Data\ op_models_gmu\ directory) The network model you open should look like this. The network model that you are looking into has been imported from real network device configurations (Cisco routers). You might be able to click with the right mouse button over Boston_Engr router and select Post Import Operations -> View Collected Output Files to see one of these configurations. If it doesn t work and you want to see it, remind me and I will bring one configuration file to demo it to you during one of the upcoming classes. Close configuration file window. Page 2 of 17
Delay performance across different links in the model can be assigned by modifying model attributes. Click with the right mouse button on top of one of the links in the model and choose Edit Attributes (Advanced). The pop up window similar to the one bellow will pop up. Look for delay attribute and instead distance based type 0.05 to specify to specify 50ms delay for this link. Click ok to close the window. Any attribute in the model can be changed this way, but you would need to get familiarized with the attribute hierarchy for individual objects. There are other more automated ways to change attributes and model characteristics, however, we will not explore these for now. Commercial, non-academic version of this software has many additional capabilities. Go to File -> Close to close this project as this was just a brief intro to the tool. Click Don t Save since we don t need to save it. Let me know if you have any questions before we continue. Page 3 of 17
NOTE: Before you start look into Appendix A, at the end of this document, to verify or setup modeling repository. Simulations will not work without it. This might be enabled by default but it is good we verify it for the first time. 3) We will now open another project. Go to File Open and open project named Voice project located in C:\Riverbed EDU\17.5.A\models\std\example_networks\voice.project. You can get to it easier if you click on a Windows icon circled in red on the following image. This batten can be used to switch between Windows and Riverbed specific browsing modes. 4) Change scenario to SIP by selecting Scenarios > Switch to Scenario -> SIP Page 4 of 17
You will now see SIP scenario as shown in the following screen capture. SIP Scenario Stop now, after SIP Scenario is opened, and wait for other classmates to get to the same point. You may take a quick break and come back. I will guide you through the remaining part of the lab manual from this point on. Let s continue together (High level instructions are also available to remind you once you start doing the lab on your own.) 5) Visit Application Config and Profile Config objects within the SIP scenario to observe codecs being assigned to the VoIP application and to define how VoIP profile configuration is executed (call initiations and durations). You can do this by clicking with the right mouse button on the top of these objects and by choosing Edit Attributes. We will start by getting into Application Config object first. Once you click with the right mouse button, navigate to app all the way till you get to the codec selection (Application Definitions -> Voice - IP Telephony -> Description -> Voice). This is the place you will use later on to change codecs and complete your lab assignment. See the image on the left. Close all open application definition windows for now. The next step is to inspect configuration of the Profile Config object. We will use this object to define call behavior characteristics. Set the call duration to 300 seconds (constant) and inter-call time to 300sec (constant). See the image on the right. Navigate to Profile Configuration -> Voice - IP Telephony -> Applications and use the Duration attribute to set call durations and Repeatability attributes to set call execution patterns. Application Config Profile Config Page 5 of 17
6) Cloud object allows you change network performance parameters - packet loss, delay (Change ip cloud delay attribute to 0.02 constant (20ms) as shown on the following image). You can get to these attributes by clicking the right mouse button on the top of the IP Cloud object and choosing Edit Attributes. 7) Navigate to Performance Matrices -> Packet Discard Ratio for packet loss specification and to Performance Matrices -> Packet Latency to specify network packet latency within the network. 8) You will now need to go to the DES tab in the top menu and define what SIP and voice-related statistics will be collected during the simulation. (You will find these under Node Statistics including MOS, jitter, etc). Question 1: How do you define MOS? Does MOS capture and measure SIP signaling performance? Question 2: How do you define GoS? Page 6 of 17
9) Follow these screenshots to choose statistics the following statistics (all voice application, calling and called party statistics; all SIP UAC and UAS statistics; and all RTP statistics): DES -> Choose Individual Statistics Page 7 of 17
10) Before you start running your simulation let s see how these applications and profiles get assigned to the VoIP terminals (UA clients). Open the caller location by double clicking on San Francisco subnet. Right mouse click on the caller and chose Edit Attributes once again. On the caller side you assign VoIP (or any other app) by going into Applications -> Application: Supported Profiles section. The Voice IP Telephony profile has already been assigned. Once here we will look into SIP specific attributes and find the Proxy Server specification. This is the server that has already been added to the scenario and you will look into just a bit later. You can close this window and move to the callee side now. Return to the parent subnet by clicking with the right mouse button somewhere on the project screen (but not on any of the objects in the window) and selecting Go to Parent Subnet option from the list. Once you are in the parent subnet double click on the Pittsburgh subnet to visit callee side of the scenario. Once there, right mouse click on the callee and chose Edit Attributes once again. On the callee side you assign VoIP (or any other app) by going into Applications -> Application: Supported Services section. The Voice IP Telephony application has already been assigned. You can close this window and move to the proxy server side. You probably noticed, VoIP application is defined in one direction only. You could repeat and define it for the opposite direction but you will not need to do it for this lab. Page 8 of 17
Jump back to the parent subnet and go to the Dallas subnet. There is a proxy_server object there. The only parameter you will need to change for the second part of the lab is Maximum Simultaneous Calls. We will leave it for now to be unlimited. Class the proxy_server attribute window and move to the parent subnet. 11) Let s now run Discrete Event Simulation (DES) and collect results we will be analyzing. Go to DES -> Configure/Run Discrete Event Simulation... Page 9 of 17
After DES completes look at those collected results/statistics. Go to DES -> Results -> View Results Choose DES Graphs as shown (browse and select node-level statistics). Go to San Francisco and expand the three to the left to get to MOS checkbox. Select the checkbox and you should see he graph in the panel to the right of the window. Now change the Presentation from As Is to average and click Show button to open the panel. Click with the right mouse button in the graph and choose Export Graph Data to Spreadsheet to obtain all values you will use for codec MOS analysis. Use these values to calculate mean, min, max, 95 th percentile and Standard Deviation. Page 10 of 17
You can now start experimenting and analyzing performance of standard voice codecs using the different network characteristics. Here are few examples: - Change codec to g.711 and no compress/decompress delay (Observe difference in MOS). - Continue changing codecs (G.711 Silence, G.726 32K, G.726 32K Silence, G.728 16K, G.723.1 5.3K, G.729A, G.729A Silence) and observe different MOS values. - Change network delay (210ms; constant) and observe MOS impact. - Bring delay back to 20ms, change network packet loss to 9%, and observe MOS impact. RECORD your results in the table listing codes and observed results. Summarize (don t just list these results but explain your observations) codec performance under different scenarios and identify what is the best and worst performing codec (for these three network condition) as part of your lab report. Once you are done with the first part: 12) Add there more callers and callees at both ends/subnets (copy and paste those you already have in this scenario). Use standard Windows copy/paste methods to complete this action. Click to place the node and then connect it to the router. - Set Maximum Simultaneous Calls attribute on the proxy server to 2 (Observe DES results for proxy server - calls that are blocked, calls that are completed, etc.). Change call arrival and duration distributions from constant to exponential and look into DES proxy server results. Calculate GoS taking simulation results from this scenario. Page 11 of 17
13) There is one more activity you will have to complete in this lab and that is latency comparison for H.323 direct endpoint call signaling (peer-to-peer) and the gate-keeper routed call signaling methods. - You will use the same Voice project but different scenario. Go to Scenarios -> Switch to Scenario -> H323. There are two subnets in this scenario, San Francisco and Pittsburgh. You probably already think there is a caller in one of them and a callee in the other one. And you are certainly right. Well, stop for a second. There is no SIP server (and it shouldn t be there with H.323 terminals) and there is no Gatekeeper. You are right again. First of all you will experiment with the peer-to-peer call setup between two H.323 terminals, collect statics and then add H.323 Gatekeeper connecting it to the router in San Francisco (as we mentioned during the first part of the lab you would most likely end up connecting these systems to the Ethernet switch but for simplicity we have direct connection to the router once again. - Make sure to collect Setup Time statistics for H323 before you start running the simulation. - Run DES and export call setup time statistics for San Francisco caller. You will compare these to those obtained by running the scenario with the gatekeeper-routed call setup. 14) Add one Gatekeeper within San Francisco subnet. - Go to San Francisco scenario. - Chose Topology -> Open Object Palette from the top menu. A new window will pop up. Page 12 of 17
- Type gatekeeper in the Search by name: box and click Find Next button. Double click on H323_Gatekeeper object and move the mouse to the subnet window to add gatekeeper to the model. Once in the subnet click with the left mouse button and then Esc button on the keyboard to stop adding additional gatekeeper objects. You should see the gatekeeper added to the subnet. Change the name of this object from node_0 to Gatekeeper. You can do this by clicking the right mouse button on the top of the object and choosing the Set Name option. - You will now need to connect gatekeeper to the router by adding the link between two objects. You can do this by copying and pasting link model that already connects workstation to the router. - Change delay on this link from Distance Based to 245ms (0.245). Make sure you click OK once you are done with the change. - Change call signaling mode on a caller workstation from Direct Endpoint Call Signaling to Gatekeeperrouted Signaling for H323 specific attributes on this workstation. Page 13 of 17
15) You are now ready to run another DES simulation. - Go back to the parent scenario and run DES one more time. - Export caller s setup time statistics Compare delay statistics from both DES runs and provide your observations to me. END Page 14 of 17
Appendix A Making sure model repositories are set correctly for Discrete Event Simulation. The release of the OPNET software you are working on is probably newer, thus, some of the windows will look different. - Click on Edit and select Preferences. Page 15 of 17
- Type repos and click the Find button. Network Simulation Repository will be selected. Open the Value field for this preference. Click on the Insert button stdmod valu: Page 16 of 17
And click OK twice. Page 17 of 17