...iLab Grading Rubric Category | Points | Description | Project 4-1: Compute RF Behavior (page 152) 25 points * Step 4 Calculation * Step 11 Calculation * Step 15 Calculation * Step 17 Calculation * Step 19 Calculation * Summary of RF Behavior | 2 2 2 2 2 15 | Calculate each of the five measurements in the project.In your own words, summarize what you have learned concerning RF behavior from your calculations, readings, and research. | Case Project 4-3: MIMO (page 156) 25 points * Introduction * Advantages/Disadvantages * Applications * Future | 55510 | Discuss MIMO antenna technology, including history, advantages, disadvantages, current applications, and future. | Total | 50 | A quality paper will meet or exceed all of the above requirements. | Deliverables IMPORTANT: A report template is provided beginning on the next page, and you must use it to submit your assignment. Before submitting your assignment, delete the pages containing the instruction and rubric. Your submitted assignment should begin with the title page and only contain your report. Don’t forget to place your name, your professor’s name, and the date on the title page. Week 2 iLab Report Francisco Santoya DeVry University NETW360: Wireless Technologies and Services COMPUTE RF BEHAVIOR AND MIMO Submitted to: Professor: Jalinous Date: 9/11/13 Compute RF Behavior Calculate each of the five measurements in the project by performing every...
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...NETW320 -- Converged Networks with Lab Lab # 2 Title: Silent Suppression – Cont. Procedure Start OPNET Modeler Open the scenario 1. Select File/Open. 2. Select Project. 3. Open your f: drive. (Note: If you don’t see drive F: listed, you click on “My Computer or “Computer” first.) 4. Open your op_models directory. 5. Open your NETW 320 directory. 6. Open the Lab1_Silent.project folder. 7. Click on Lab1_Silent.prj. 8. Click OK. The project should open. 9. Choose Scenarios > Switch to Scenario > Silent_Suppression. Results analysis 1. We are now ready to look at the results. From the tool bar, select DES > Results > View Results. 2. The Results Browser will appear. You may have to expand the items in the top left panel and click on them to get Global Statistics to appear in the bottom left panel. 3. Expand Global Statistics and select the following (4) statistics: Expand Select DB Query Response Time (sec) HTTP Page Response Time (sec) E-mail Download Response Time (sec) Voice Packet End-to-End Delay (sec) 4. You can adjust the size of the panels as you wish by hovering the cursor over the panel border until it changes to the adjust line cursor, and then hold the left mouse button to set the panel size. 5. Change the view from As Is to Time Average using the dropdown menu on the lower right-hand side. Remember, your results may not be exactly the same but they should be very similar. 6. Click Show. A graph similar...
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...NETW320 -- Converged Networks with Lab Lab #3 Title: IPv4 TOS and Router Queuing Objectives In this lab, you will work with an intranet for an organization that will encompass four different site locations in different cities. The subnets of these locations will be connected by a backbone IP network. The organization will be using a converged network that allows data and real-time voice traffic to traverse the same packet-switched network. The data traffic will consist of FTP (file transfer protocol) and email traffic and the voice traffic will be a VoIP (Voice over Internet Protocol) implementation. You will experiment with various router queuing policies to see how routers within a TCP/IP network can be utilized to support QoS (Quality of Service) within a converged network that is based on TCP/IP. Explanation and Background Traditional voice and data applications have been kept on separate networks. The voice traffic is confined to a circuit-switched network while data traffic is on a packet-switched network. Often, businesses keep these networks in separate rooms, or on different floors, within buildings that they own or lease (and many still do). This requires a lot of additional space and technical manpower to maintain these two distinct infrastructures. Today’s networks call for the convergence of these circuit-switching and packet-switching networks, such that voice and data traffic will traverse a common network based on packet switching. A common WAN technology...
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...Course Project (80 points) QoS Design and Implementation 1. Objectives As a network engineer working for a Service Provider, design and implement QoS on a converged network. For this project it is assumed that the converged network offers triple play services and has the following traffic: VOIP, High Definition TV (HDTV), Video on Demand (VOD) and Internet. Note 1: The project is based on the experience acquired by the students in the previous labs, hence the main sources to complete the project are the labs of previous weeks, so it is important that the students review the lab instructions of the labs that have already been completed. Note 2: The students will have access to the same lab equipment of Skillsoft as the previous weeks. The students will need to access the iLab to complete the project. 2. Project Requirements Part A: Design In this part the student should provide the QoS design. A1. It is assumed that the traffic sources for HDTV, VOD, VOIP and Internet are connected to switch NYCORE1 to four different ports. The following are the requirements: (10 points) * Assign a switch port to each type of traffic. * Assign a COS to each type of traffic. The COS marking has 8 classes, from 0 to 7, assuming 0 is the lowest priority and 7 is the highest. Decide which class each traffic should have, based on the importance of the traffic. * Provide your answers in the form of the table below: Traffic Type | Switch Port Number | COS | VOIP |...
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...NETW320 Week 5 and week 6 – Asterisk* Lab Here You can see that the address is set and that the Ports are now active. And I can get a line for 3 seconds Using my windows phone but it doesn’t stay connected. I have now been working on getting tested with Chas from class This is the coding that I used to set up my Config line. I am still working on the configuration setup.This was taken out of the Red Hat Book. John's extensions.conf [general] static=no writeprotect=no autofallthrough=yes clearglobalvars=yes priorityjumping=no [globals] ; Incoming calls to these numbers will be routed to the specified peer. peer-5555551234 = SIP/mango peer-5555554321 = SIP/mangosteen ; Incoming calls to these numbers will be routed to the specified context. context-8775551234 = DISA-in,s,1 ; These names will override whatever is sent as the Caller ID Name for incoming calls. callerid-5555556789 = Joe Blow ; These are the peers outgoing calls may be routed through, in order. If one is unreachable, Asterisk will attempt to use the next one. outgoing-1 = VoIPmsUS3 outgoing-2 = Callcentric ; This is the password for use with DISA. ...
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...RTP – short for Real-time Transport Protocol defines a standard packet format for delivering audio and video over the internet. It is defined in RFC 1889. It was developed by the Audio Video Transport Working group and was first published in 1996. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and web-based push-to-talk features. RTP is an end-to-end transfer of data in real-time. What this means is that data is sent from the server to the client(s) and is in actual time. Imagine something like Internet radio which is audio only, but there should be few lapses in the transmission since the data is usually buffered. A buffer is when data is stored in memory and played from memory. The buffer may allow for a few seconds of stored information before playing. Some applications can allow for setting the buffer amount. RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP carries the media streams (e.g., audio and video), RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization of multiple streams. RTP is originated and received on even port numbers and the associated RTCP communication uses the next higher odd port number. RTP is one of the foundations of VoIP and it is used in conjunction with SIP which assists in setting up the connections across the network. The Real-Time Transport Protocol...
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...NETW320 -- Converged Networks with Lab Lab #4 Title: IPv4 TOS and Router Queuing – Cont. Procedure Start OPNET IT Guru Open the scenario 1. Select File/Open. 2. Select Computer or My Computer (depending on your O/S). You may also need to allow Citrix access to your computer. 3. On your F: drive, open the op_models and then open the NETW320 folder. 4. Open the Lab2_RouterTOS.project 5. Click on Lab2_RouterTOS.prj 6. Click Open. The project should open. 7. Choose Scenarios > Switch To Scenario > FIFO Configure the Simulation Run 1. We are now ready to configure the Simulation Run. Select the Configuration/Run Discrete Event Simulation tab (the running man) from the tool bar. The following screen will open. 2. Set the Duration to 4 (if it is not set) and change hour(s) to minute(s). 3. Click Apply and Cancel. 4. Go to File > Save, to save your configuration. 5. Before we duplicate the scenarios, now would be a good time to run the first simulation to ensure we have all the configurations made correctly. Once we copy them over to the PQ and WFQ scenarios, if something is configured incorrectly, that mistake will be transferred over. 6. Select the running man icon again to bring up the Configuration/Run Discrete Event Simulation panel again and click Run. The Simulation Execution window will open and the sim will start. 7. When the Sim completes and the Close window lights, click it to end. 8. We are now ready to look at the results. From the tool...
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...Required Lab Summary Report for NETW320, Codec Selection for Campus Network (This section is worth 25% of your grade for this lab.) Write a two-page summary report and use the graphs that you generated during this lab and the tables that I gave you in the introduction to the lab to support your analysis. You are also encouraged to seek additional reference material, such as your textbook and the Internet, to support your arguments. Your two-page summary report must answer the following: Which of the codecs used in this lab would you recommend that the Campus Network Administrator implements? To answer this question, you need to consider MOS, bandwidth consumption, overall delay, and variation delay. Use the following questions to help you decide and support your answer. 1. How did particular codec implementations affect DB Query response time and e-mail-download response time? Were any of these delays significant enough to cause database users and e-mail users to become dissatisfied? 2. Compare the amount of voice packets generated by the various codecs. What effect does generating more voice packets for a particular codec have on voice quality? Why? 3. Use the overall delay that you calculated for the three codecs used in this lab to rate their delay times with the ITU-T benchmark outlined in the introduction to this lab. 4. Comment on the effect of your measured packet-delay variation for the various codecs analyzed in this lab. See the comments on packet-delay variation...
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...NETW320 Real-Time Transport Protocol Real-Time Transport Protocol, more simply known as RTP, is a two part standard of how to manage real-time transmissions of multimedia data across a network. This two part standard first contains RTP, which is responsible for data transport and flow. The second half is Real-Time Control Protocol (RTCP) which is the control portion that monitors RTP transmission, collects data and compensates for lost packets and jitter, as well as handles Quality of Service (QoS). In general RTP is run over the User Datagram Protocol (UDP), but it is capable of running over other protocols such as TCP as well. With today’s converged networks and their inclusion of VoIP and the heavy use of the Session Initiation Protocol or SIP, it is easy to understand why RTP is utilized. When you look into the relationship between RTP and SIP, you can clearly see that SIP relies on RTP. To fully understand their relation, we can break it down. First you have SIP. This protocol is responsible for setting up the connection from end to end. Once the connection is established, RTP takes over to transmit the data stream. While both SIP and RTP serve different functions all together, they rely on each other to send and receive the voice streams at either end. VoIP, however, is not the only use for RTP. It is used for a variety of audio and/or video streaming and its core design allows for the ability to support a plethora of formats, as well as the ability to incorporate...
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...Rollin Luis NETW320 Professor Antoniou Week 2 HW assignment Silent suppression is a technique used to improve bandwidth utilization of voice circuits. Through silence suppression, a mechanism senses periods of inactivity in a voice conversation and simply ceases sending data associated. Silent suppression can be enabled to monitor signals for voice activity so that when silence is detected for a specified amount of time, the application informs the Packet Voice Protocol and prevents the encoder output from being transported across the network. To help understand silence suppression technology, consider this example: Some people talk all the time. Even they have to take a breath and they sometimes listen and are quiet. On a phone call, it is quite common that there is silence in one direction of the call while there is speech carried in the other direction. The public switched telephone network does not take advantage of this condition. The PSTN opens up a path in both directions, between the speakers, and allocates 100% of the path capacity to the call, even when there is silence. Silent suppression is the application of this IP network principle to VoIP calls. When there is silence, you don't send voice packets full of silence. Silence suppression can save bandwidth, especially on IP trunks. The savings can be 40% to 50%. The packet-sending phone or gateway implements the VAD function in the codec. When you think of silent suppression think of it as speech door. It opens...
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...NETW320, Patrick Price 5/18/2014 Lab #2, Silent Suppression Lab Report 1. On the Results Browser, make sure you are on Current Project so you have both sets of results. Expand DB Query and Select DB Query Response Time (sec). Hit the Show button. Zoom into the last half portion of the graph for better granularity and to avoid start up oscillation time to stabilize. Copy and label this graph to your lab report and answer the following: 1. Which run has a better (lower) DB Query Response time? This shows the silent suppression has a lower db time. 2. In regard to your answer to part a, approximate how much faster (in seconds or milliseconds) of a response time the better scenario has. Its .22 seconds faster. 2. Expand E-mail and select Download Response Time (sec). Select Show and zoom into the last half portion of the graph for better granularity and to avoid start up oscillation time to stabilize. Copy and label this graph to your lab report and answer the following: 3. Which run has a better (lower) e-mail Download Response time? Again it’s the silent suppression that had a email response. 4. In regard to your answer to part a, approximate how much faster (in seconds or milliseconds) of a response time the better scenario has. It was .4 seconds faster. 3. Expand HTTP and s elect Page Response Time (sec). Select Show and zoom into the last half portion of the graph for better granularity and to avoid start up oscillation time to stabilize...
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