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Simulation and calculation of throughput for a TCP connection (using ns2 Simulator) 2. Simulation and calculation of throughput for a for a star connected network with 2 TCP and 1 UDP connection (using ns2 Simulator) 3. 4. 5. 6. 7. 8. 9. Simulation of Local Area Network (using ns2 Simulator) Simulation of Mobile Adhoc Network and comparison of the performance of Routing Protocol (DSDV, AODV&DSR) (using ns2 Simulator) Simulation of a Bluetooth system (using System View software) Simulation of an IEEE 802.11a system (using System View software) Assembly of GSM set up & Real Time study of GSM 07.05 & 07.07 AT commands (such as network registration, call control, call setting etc.) Study and Implementation of Auto Dial, Call Forwarding in IP Telephony Configuration of an IP Phone and Implementation of Class of Service (COS) and Class of Restriction (COR)
10. Study of the working principle of Address Resolution Protocol(ARP) 11. Simulation of an Ethernet Network and to study its performance under different scenarios 12. Study Of Resource Reservation Protocol (RSVP) as a part of the Integrated Services Approach for providing Quality Of Service (QoS) to Individual Applications or Flows.
OPTIONAL EXPERIMENTS: 1. Network Security: Exploitation of known flaw in send mail and implementation of a firewall in such a way that would make this flaw unusable 2. Construction of six node TCP / IP Network and analysis of the use of RIP Vs OSPF as the Active Routing Protocol 3. Analysis of the signal strength and throughput at several locations in a wireless LAN 4. Study and analysis of the working of DHCP & NAT in a network 5. Programming with Cryptographic libraries: Encryption and Decryption 6. Study of Advanced Encryption standard (AES) 7. Study of the concept of Cellular System Design (such as Frequency Reuse, Sectorization, and Frequency Channel Assignment) and investigation of the effect of Demographic, Traffic, Terrain on Cell Planning and to expose the students tool used for cell-planning 8. Setting up a Virtual Local Area Network for voice IP
LAB INSTRUCTIONS FOR CARRYING OUT PRACTICAL ON SIMULATION AND CALCULATION OF THROUGHPUT FOR A TCP CONNECTION (USING NS2 SIMULATOR)
AIM: Simulation and calculation of throughput for a TCP connection (using NS2
Simulator)
0
Source
Link-1
Link-2
2
Destination
Both Link-I and Link-II are duplex lines with following parameters. LINKS Link-I Link-II Do the following. Create a TCP connection with FTP application running over it at node 0 for the destination at node 2. Use the Reno variant of TCP. Schedule as follows TCP connection: 0.5sec to 50.5 second. Simulation time: 0 sec to 51 second. Run the simulation and calculate the throughput. Data rate 100 Mbps 100 Kbps Latency 10 ms 30 ms
3 4
0 6
Figure 1.2
As shown in the above network (Figure1.2), create 5 FTP connections that start at random: the starting time is uniformly distributed between 0 and 7 sec.The whole simulation duration is 10 seconds. Create links with delay that is chosen at random, uniformly distributed between 1ms and 5 ms. In addition to the standard trace outputs, create a file named win that will contain the evolution of the window size of all connection at a granularity of 0.03 sec.
PROCEDURE:
1. Develop the ns2 code for the given experiment as follows Create the simulator instance Open the trace and NAM file Define the finish procedure Create bottleneck and destination nodes and link between them. Create a random generator for starting the ftp and for bottleneck link delays Create the TCP connection and attach FTP to them Schedule events for the FTP agents. Include the code to plot the TCP window size.
RESULTS:
LAB INSTRUCTIONS FOR CARRYING OUT PRACTICAL ON SIMULATION AND CALCULATION OF THROUGHPUT FOR A FOR A STAR CONNECTED NETWORK WITH 2 TCP AND 1 UDP CONNECTION (USING NS2 SIMULATOR)
AIM: Simulation and calculation of throughput for a for a star connected network with
2 TCP and 1 UDP connection (using NS2 Simulator)
1 5
2 Mbps 10 ms RED
7 9
20 Mbps 10 ms DropTail Buffer Size=3 pkts
6 0
3 4 8 2
The above network with star topology has 10 nodes (node 0 through 9). There are two FTP applications running over TCP at nodes n (0), n (1) & one CBR running over UDP at node n (2). The destinations of node 0, 1, 2 are 3, 4, and 5 respectively. All the links are full duplex links. Schedule: All the TCP connections start at 0.5 second and stop at 10 second. The UDP connection start at 1 second and stops at 10 second. Simulation time is from 0.5 sec to 10.5 sec. Do the following: Develop the necessary ns2 code and run it. Analyze the trace file and plot graphically 1. Number of pkts received 2. Number of pkts dropped 3. Throughput of all TCP and UDP connection. Look at the trace file and manually calculate the throughput for all TCP and UDP connection and compare the results to that of graphical methods. Calculate the average packet delay from source to destination ( for all connections) and percentage of packet dropped over entire simulation time Change the buffer size of the links to the hub routers to 5(instead of 3) and run the simulation and repeat the above calculation and analysis.
SOFTWARE REQUIRED:
NS2 simulator (ns 2.29/2.30) (under Linux), Trace Graph 3.02
PROCEDURE:
1. Develop the ns2 code as follows. Create a Simulator Object Define different colors for data flows Open the nam/ trace file Define a finish procedure Create 10 nodes from n0 through n9 Establish duplex links Define node position for creating the star topology as in figure Create TCP agent and then run ftp over it for all three sources and sinks. Create UDP connection and run CBR over in the appropriate node. Define Start time and Stop time for TCP agents Define End time for Simulation. 2. Run the program and debug it properly 3. Once the simulation completed successfully, analyze the *.tr file and calculate the throughput for all TCP and UDP connection. 4. Open the *.tr file in trace graph and make the necessary analysis. 5. Save the graph and simulation results
RESULTS:
3D graph relating to number of packets received. 3D graph number of packets dropped. 2-D graph for throughput of all the TCP and UDP connection Simulation results (Through trace graph)
LAB INSTRUCTIONS FOR CARRYING OUT PRACTICAL ON SIMULATION OF LOCAL AREA NETWORK (USING NS2 SIMULATOR)
AIM:
2 Mbps, 10 ms
LAN (0.5Mb, 40ms, Drop Tail Queue, MAC, CSMA/CD 300 Kbps 100 ms
2 1
2 Mbps, 10 ms
Write NS-2 code to create the above topology Buffer limit of the link N2-N3 is 20 Other links have default Buffer limit. Establish a TCP connection between nodes 0 and 4 which will be active from 1.0 sec to 124.0 sec with TCP packet size 500 bytes. Establish an UDP connection between nodes 1 and 5 which will remain active for duration 1.5 sec to 124.5sec with a data rate of 10Kbps and of packet size 1000 bytes. Analyze the trace file and plot the following o Throughput o Average packet delay o Packets dropped o Instantaneous size of the TCP window at interval of 0.1 sec.
SOFTWARE REQUIRED:
NS2 SIMULATOR (NS2.29/2.30) under Linux, Trace Graph, XGRAPH feature available within ns2 simulator.
PROCEDURE:
1. The ns2 code can be developed by using the following procedures. (The procedures given below are just a reference. Students are welcome to modify it any manner with their programming ability.) Create a Simulator Object Define different colors for different data flows Open the Trace file Open the NAM file Define a finish procedure Create the 6 nodes
Setup a TCP connection Setup a FTP over TCP Setup an UDP connection Setup a CBR over UDP Schedule the event 2. Run the program and debug it properly 3. Once the simulation completed successfully, analyze the *.tr file and calculate the throughput for all TCP and UDP connection, Average Packet delay etc.. 4. Open the *.tr file in trace graph and make the necessary analysis. 5. Save the graph and simulation results
OSERVATION
1. 2. 3. 4. 5. 6. Graph of packets received at all nodes Graph of packets dropped at all nodes Throughput of the packet received between node 0 and 4 Throughput of the packet received between 1 and 5 Plot of TCP Window by using X-graph (as shown below) Detailed simulation results
DEPARTMENT
LAB INSTRUCTIONS FOR CARRYING OUT PRACTICAL ON SIMULATION OF MOBILE ADHOC NETWORK AND COMPARISION OF THE PERFORMANCE OF ROUTING PROTOCOL (DSDV, AODV&DSR) (USING NS2 SIMULATOR)
AIM:
Simulation of Mobile Adhoc Network and comparison of the performance of routing Protocol (DSDV, AODV&DSR) (using NS2 Simulator)
PROBLEM Simulate a Mobile Adhoc Network consisting of 10 nodes randomly placed in an area of 1.5 km x 0.8 km. The entire nodes move in a random fashion and also communication among them is random. Assuming the routing protocol to be DSDV initially, analyze the performance of the network in terms of throughput, delay, TCP window size etc Replace the routing protocol DSDV into AODV, DSR & repeat the above analysis. Compare the obtained results. Take the simulation time to be 100 second.
SOFTWARE REQUIRED:
NS2 SIMULATOR (NS2.29/2.30) under Linux, Trace Graph, and XGRAPH feature available within ns2 simulator.
PROCEDURE:
Specify the basic parameters for simulations i.e provide information for different layers Define the parameters which are used in the configuration of the mobile nodes. Create the desired number of nodes with the above mentioned parameters. Use the following script to generate the desired number of nodes and the connection pattern ns cbrgen.tcl [-type cbr/tcp ] [ -nn nodes ] [ -send node ] [ -mc connections] [ -rate rate ] For example: ns cbrgen.tcl type tcp nn 10 send 0 mc 15 rate 4 > filename with path As the placement of the node as well as the movement of the nodes are random, use ./setdest to solve the purpose. (For example ./setdest v 2 n 10 S 2 m 5.0 M 10.0 t 100 P 1 p 5.0 x 1000 y 800 > filename with path) Write the code to print window size Schedule the different events. Run the simulation successfully. Analyze the results in terms of throughput, delay and window size. Change the routing protocol from DSDV to AODV and then to DSR. Simulate it again and repeat the analysis. Compare all the results obtained.
DEPARTMENT OF
LAB INSTRUCTIONS FOR CARRYING OUT PRACTICAL ON SIMULATION OF A BLUETOOTH SYSTEM (USING SYSTEMVIEW SOFTWARE)
AIM:
Simulate a Bluetooth System by using System view software with the following parameters.
THEORY: PROCEDURE:
Open the workspace and make connection as per the block diagram shown in the Figure 2. FH frequency command generation: Tokens 1, 2, 3 and 4 generate the FH frequency commands. Token 2 is a uniform random number generator set over the range 0 to 79e6 (79 frequencies @ 1 MHz spacing). This token is then sampled at the hop rate, 1600 hops/s.Taking the integer part generates the hop frequency commands. Data frequency command generation: The 1 MBPS data source (token 6) is passed through a Gaussian LPF (token 7) set for a BT product of 0.5. The resulting output is passed through a Gain set to 140000 (token 8). This produces the desired FM modulation index of 0.28. GFSK Modulation: The FH hop command signal is added to the data signal, and the composite drives the VCO modulator (token 0). Another method would be to separately FM modulate the data and the FH signal, and then mix the two. The VCO is set to a nominal frequency of 10.7 MHz. When mixed by the FH demodulation signal, all hops are collapsed to this IF frequency. TDD Control: The TDD control is accomplished by blanking out the modulator. A square wave signal (token 10) is set to a 1.25 msec period. It is on for 625 sec and then off for 625 sec.
FH Demodulation: The FH hop commands are used to drive a VCO (token 14). The output of the VCO is mixed with the received FH signal. The difference frequency is at the 10.7 MHz IF, and the sum frequency is eliminated by the band pass filter (token 16). IF Processing: After the FH dehop operation, the resulting signal is at 10.7 MHz. A 3 Pole Butterworth Band pass IIR (token 16) with a band pass low Fc of 9.7e+6 Hz and a high Fc of 11.7e+6 Hz is used as the standard IF filter. PLL Demodulation: A PLL (token 11) is used to directly demodulate the signal from the 10.7 MHz IF frequencies. The PLL token has an internal LPF after the phase detector to eliminate the sum frequency from the loop. The cutoff frequency is set at 4 MHz which is wide enough to pass the desired signal. The modulation gain is set at 2e6. This gives the loop response time sufficient to demodulate the signal.
Run Simulation.
Observation:
Observe the input and output waveform Overlay the input and output data signal and obtain the percentage of agreement.
LAB INSTRUCTIONS FOR CARRYING OUT PRACTICAL ON SIMULATION OF AN IEEE 802.11A SYSTEM (USING SYSTEMVIEW SOFTWARE)
PROCEDURE:
Open the workspace of the System View and create the 802.11a modulation and demodulation units as shown in Figure 1. Token 10: 36 Mbps data source which is sampled (Token 11) at once per bit. Token 12 is the [131,171] constraint length 7 convolutional encoder. The data out of the encoder is 72Mbps. Token 13 follow this, which performs the puncturing operation. The net effect is that for every 3 bits into the convolutional encoder, there are 4 bits out of the puncture token. Thus the rate is now 48 Mbps. The data is then interleaved in Token 27. The bit-to-symbol token 14 and the QMAP token 16 combine to produce the proper baseband I and Q signals. The symbol rate is 12 Msym/sec. Now assemble the data into the required packet structure. The General DeMux token 18 splits the data symbols into the appropriate segments for use by the GenMux Token 19. Figure 2 shows the parameters of the DeMux Token 18 The Gen Mux token 19 assembles the packet for the I signal (a similar discussion applies to the Q signal).
Figure 3 is the Gen Mux token parameters. It will be recognized as the packet structures resulting in 13 segments total with segments (1,2,3,5,6,7) as 48 data sub carriers per symbol; segments (2,4,8,10) carrying the sync data; segment (7) which is the center frequency set to 0,and segments (0, 13) are 0 fillers.
The sync data is controlled by a 7 stage PN source (token 16) described by the polynomial. The total is 64 carriers required by the OFDM symbol modulator. Obtain the eye diagram of the constructed packet as shown in Figure 4.
The 4 pilot carriers, which are BPSK modulated, are clearly visible, as is the null carrier in the center. The final I and Q signals are then sent to the OFDM modulator token 5. Set the OFDM modulation parameters as shown in Figure 5.
The steps just described for the modulation process are applied in reverse order to recover the original data.
LAB INSTRUCTIONS FOR CARRYING OUT PRACTICAL ON ASSEMBLY OF GSM SET UP & REAL TIME STUDY OF GSM 07.05 & 07.07 AT COMMANDS (SUCH AS NETWORK REGISTRATION, CALL CONTROL, CALL SETTING ETC.)
AIM: Assembly of GSM set up & Real Time study of GSM 07.05 & 07.07 AT commands (such as network registration, call control, call setting etc.) PROBLEM (A) Assembling GSM Setup (B) Real Time study of GSM 07.05 & 07.07 AT commands concerning Modem and Simcard information Network registration Call control and Call Setting Call information Phone book Serial Link control Message setting Storing/resoring Error Message handling
THEORY:
PROCEDURE:
1. Note down the Technical Data of the Evaluation Board (A2D/F35/C2D) 2. Note down the Technical data of F35-A-2 :TC35 Module with IMEI number 3. Note down the operating frequency of the GSM Antenna 4. Note the pin details of RS232 9 PIN SERIAL Cable 5. Assemble the GSM Setup as follows
Place Evalboard on a table. Connect the serial cable to the 9 pole com 1 or COM2 port of the PC and connect the other end of the serial cable to the 9-pin SUB-D connector of the EVAL-Board.
Pick up the A2D/F35/C2D Evaluation board and observe it carefully. On left most bottom corner, there is a four pin connector named BTMP A2D F35. Short the rightmost two pins by jumper.
Now take Siemens TC35 module. Fix it on A2D/F35/C2D Evaluation board carefully in a similar manner Fix SIM card either on internal SIM holder available on TC35 module or external holder on Evalkit. Take GSM antenna and connect with TC35 properly. Connect RJ-45 plug either on Evalboard or on A2D/F35/C2D BOARD. Communicate with the module through any terminal program such as HyperTerminal under windows. Use A2D test software for the purpose. Plug in the power supply and turn on the EVAL-Board. Start the test Software. Now press Soft On for three seconds until the RED LED is ON. Practice the AT commands relating to Modem and Simcard information, Network registration, call control and Call Setting , Call information, Phone book , Serial Link control, handling. Message setting,Storing/resoring,Error
Message
LAB INSTRUCTIONS FOR CARRYING OUT PRACTICAL ON STUDY AND IMPLEMENTATION OF AUTO DIAL, CALL FORWARDING IN IP TELEPHONY
AIM: Study and Implementation of Auto Dial, Call Forwarding in IP Telephony PROBLEM
To understand and implement the features of auto dial and automatic callback. To understand and implement call forwarding
APPARATUS:
S8300B media server, G700 media gateway, Avaya Communication Manager, Avaya Site Administration, System Administration Terminal, IP phone, Extreme 200-48 switch, patch cords.
THEORY: Autodial: This feature is similar to the speed dial feature present on a cellular phone. Let
us assume there is a certain number that the user has to dial very frequently. Instead of dialing the number every time, a single button on the IP phone can be configured to automatically dial the required number
Call forwarding all calls: Users use the Call Forwarding All Calls capability to
redirect any incoming calls to another destination. Users use a feature access code (FAC) or a Call Forward-All feature button to activate or deactivate Call Forwarding All Calls for their own telephone.
PROCEDURE:
(A) For Autodial and automatic Callback 1. Open Avaya Site Administration and log in. 2. Type in the command change station 400 and go to page 3. 3. There is a segment labeled BUTTON ASSIGNMENTS at the bottom of the screen. In the sixth slot enter autodial. The system asks for an extension. Enter 200. 4. In the fifth slot enter auto-cback.
5. Now if a called extension is already engaged in another conversation press the Auto Callback feature button. The line appearance on the phone darkens. Three beeps are heard as a confirmation tone. When the called party goes on-hook, the extension 400 gets a ring. As soon as the receiver is lifted the called party gets a ring. 6. Auto Callback can also be used if the other extension does not answer. The next time the called party finishes a call, 400 automatically gets a ring. The rest of the process is similar to the above step. 7. Instead of dialing 200 from the keypad, pressing the autodial button places a call from 400 to 200. (B) For call forwarding:
1. Open Avaya Site Administration and log in. 2. Ensure that all extensions are active and have the same COR and COS values. 3. The COS value assigned should have the Call forwarding field values set to y. 4. Type change station 400 and go to page 3. 5. Under BUTTON ASSIGNMENTS enter call-fwd in the fourth slot 6. On 400 press the CFrwd feature button and enter the extension number to which the calls have to be forwarded. 7. Now a call made from the third extension will be forwarded to the number dialed entered in the previous step. 8. Note that a call made from the second extension to 400 will encounter a busy tone. 9. The feature can be deactivated by pressing the feature button again.
LAB INSTRUCTIONS FOR CARRYING OUT PRACTICAL ON CONFIGURATION OF AN IP PHONE AND IMPLEMENTATION OF CLASS OF SERVICE (COS) AND CLASS OF RESTRICTION (COR)
PROBLEM
1. To configure an IP Phone.
2. To understand class of service and class of restriction and assign the two for a particular station
APPARATUS:
S8300 media server, G700 media gateway, Avaya Communication Manager, Avaya Site Administration, System Administration Terminal, Extreme 200-48 switch, cross cable, patch cords, IP phones (2).
THEORY: Class of restriction: You can use CORs to restrict communication between point A
and point B. For example, a user tries to establish a communication path between point A and point B. The system checks whether the CORs have permission to communicate with one another. If the CORs have permission, the system completes the call. If the CORs do not have permission, the system does not complete the call. You control the level of restriction that the COR provides. CORs also have other applications. You can apply administration settings to a COR, and then assign that COR to objects or facilities in the system. This use of CORs makes it easier to administer functions across a wide range of objects. To set up a COR, you select a COR number. You can assign a name for the COR that reflects either the purpose or the members of the COR. You then use the Class of Restriction screens to select what restrictions, if any, apply to the COR. After you set up a COR, you assign the COR number to objects on your system. When you administer your system, the best strategy is to assign CORs to similar groups or objects. For example, you might create a unique COR for each type of user or facility, such as: Call center agents Account executives Supervisors
To enhance your system security, you can: Assign a separate COR to incoming trunk groups and outgoing trunk groups, and then restrict calls between the two groups. Set appropriate calling party restrictions and Facility Restriction Levels (FRLs) to limit the calling permissions as much as possible. You can restrict calls from one COR to another COR to prohibit user access to specific telephones. The system routes restricted calls to intercept tone.
Class of service: Class of service is used to allow or deny user access to certain features such as automatic callback and call forwarding. The difference between COR and COS is that while the former is used to define restrictions that apply when a user places or receives a call, the latter allows or denies access to certain features.
PROCEDURE: (A) Configuration of IP Phone: 1. Take the 4610 IP phone with extension number 400 and press the buttons in the given order: Hold73738# (73738 actually spells out RESET on the keypad). 2. Once these keys are pressed a question appears on the display: Reset values? Press # to execute the command. Press # again to reset the phone. 3. The phone restarts and initializes. Wait till you see an option on the display which says * to program. At this point press the * key on the keypad. 4. Enter the IP address of the phone, using the * key in place of the . The IP addresses of the 4610 is 192.168.1.20 Press # to continue 5. The CallSv (call server) IP is 192.168.1.10, this is the address of the S8300 media server which is the call controller. Press # to continue 6. The CallSvPort has a default value of 1719, which is a UDP (user datagram protocol) port number. This should not be changed. Press # to continue 7. The call server acts as the router in this case. Hence the IP address of the router is the same as that of the call server. Press # to continue 8. The subnet mask should be given as 255.255.255.0, Press # to continue 9. There is no need for a file server as we are manually configuring the phones. This value should not be changed. Press # to continue 10. The 802.1Q default value has been set as auto. Do not change it. Press # to continue 11. The default VLAN ID is 0. This should not be changed. Press # to continue 12.A question appears now, Save new values? Press # 13.Wait for the phone to restart and initialize. Wait till the phone asks for an extension number. This should be given as 400. Press # to continue 14.The password is 1234. Enter it and press # 15.The display of the phone now shows the phones extension numbers, date, time etc. The phone is now configured for use.
(B): FOR COR (Class of Restriction) AND COS (Class of Service) 1. Open Avaya Site Administration and log in. 2. Make sure that all the extensions are active. 3. To view all the existing CORs type list cor. 4. To view a COR (in this case 2) type display cor 2. 5. Type change station 400 and assign the COR number in the COR field. 6. Press F3 to submit the value. Now restrict calls between two CORs 7. Assign cor 4 to any other phone by the method described above. 8. Type change cor 2 and go to page 3. Change the y against 4 in the first column to n and submit the values. 9. Now any call between extension 400 and the extension with COR 4 will not be allowed. 10. Type display cos. All possible COSs with various combinations of features appear on the screen. These values can be assigned by filling in the cos field after typing change extension 400.
LAB INSTRUCTIONS FOR CARRYING OUT PRACTICAL ON STUDY OF THE WORKING PRINCIPLE OF ADDRESS RESOLUTION PROTOCOL
THEORY:
The Internet Protocol (IP) uses the address resolution protocol (ARP) to map IP network addresses to the hardware (media access sub layer (MAC)) addresses used by the data link protocol. The ARP protocol operates between the network layer and the data link layer in the Open System Interconnection (OSI) model. The phrase address resolution refers to the process of finding a MAC address of a host (computer) on a network. The address is resolved using a protocol in which a short frame (data link layer packet) is broadcast on the local network by the host attempting to transmit data (client). The server on the receiving end processes the frame. The address resolution procedure is completed when the client receives from the server, a response containing the servers address. The hardware address that is sent back to the host is known as the Medium Access Control (MAC) address. Every hardware devices MAC address can be found on the network interface card (NIC), which is located inside the host. The MAC address is hard coded, which means that it cannot (usually) be altered by software. In this lab you should discover the importance of routing between two different networks (subnets) when ARPing an address. You will send packets from a client computer to a server computer using the ping. The ping program is a tool available on most operating systems that allows you to test if another host is up and active using the ICMP echo protocol. According to standards all computers running the IP protocol are required to have an ICMP echo server and for that server to be up and running at all times. You will send a packet to a computer on the same subnet and a different subnet through a router. Your main observation should determine the MAC address when the two different hosts communicate, what address is returned from the receiver of the packets? Will the addresses be what you expect or something different? You will also learn how the ARP protocol works in establishing communication. Note, if the hosts are on two different networks, the IP (wrapped in packet) must be known in order to get the MAC (wrapped in frame) address. There are four types of ARP messages, which may be sent by the ARP protocol. These are identified by four values in the "operation" field of an ARP message. The types of messages are: 1 2 3 4 1. ARP request 2. ARP reply 3. RARP request 4. RARP reply
PROCEDURE:
In this experiment you will observe the sending and confirmation of data frames that contain the MAC address. You have three terminals (two of them are in the same subnet, the third one is in different subnet; gateway connects both of the subnets). You have to capture traffic and analyze information exchange between these terminals. Your group will analyze what exactly is going on behind the scenes of the basic ping command. 6. Go to the PC1 labeled (192.168.60.100). 7. Run Ethereal Traffic Analyzer (type ethereal in the command line of terminal). 8. Go to menu and choose Capture Start. Choose eth0 interface and press OK. The program began to capture the traffic. 9. Type in the command ping on the command prompt of terminal with the IP address of 192.168.60.5 and send 10 packets to the remote host. ping 192.168.60.5 c 10 10. Go to the LAN Decoder and hit escape once you see the traffic being recorded on the network. 11. Find the information about ARP Request and ARP Reply packets. Info section of packets description should contain the following: ARP Request: Who has 192.168.60.5? Tell 192.168.60.100 ARP Reply: 192.168.60.5 is at <MAC address of destination> If you type arp in the command line, you will see that the terminal stored the information about destination MAC address in its ARP cash. 12. Find ICMP Request and ICMP Reply (ICMP is the protocol used to exchange ping messages between two hosts). 13. Look at the detailed information about each of four packets and fill the information in Table 1(a) 14. Record ping average response time in the table below (Table 1 (d)) 15. Repeat steps 3-9, but use IP 192.168.60.2 (another node that is acting as the default gateway) and fill in Table 1 (b) 16. Repeat steps 3-9, but use IP 192.168.24.100 (node located in different subnet) and fill in table 1c. Clean arp cash before this experiment (type arp d 192.168.60.2 in command line).
LAB INSTRUCTIONS FOR CARRYING OUT PRACTICAL ON SIMULATION OF ETHERNET NETWORK AND TO STUDY ITS PERFORMANCE UNDER DIFFERENT SCENARIOS
AIM: Simulation of an Ethernet Network and to study its performance under different
scenarios To demonstrate the operation of Ethernet Network its performance under different scenarios. & to examine
THEORY:
The Ethernet is a working example of the more general Carrier Sense, Multiple Access with Collision Detect (CSMA/CD) local area network technology. The Ethernet is a multiple-access network, meaning that a set of nodes sends and receives frames over a shared link. The carrier sense in CSMA/CD means that all the nodes can distinguish between an idle and a busy link. The collision detect means that a node listens as it transmits and can therefore detect when a frame it is transmitting has interfered (collided) with a frame transmitted by another node. The Ethernet is said to be a 1persistent protocol because an adaptor with a frame to send transmits with probability 1 whenever a busy line goes idle. In this experiment you will set up an Ethernet with 14 nodes connected via a coaxial link in a bus topology. The coaxial link is operating at a data rate of 10 Mbps. You will study how the throughput of the network is affected by the network load as well as the size of the packets.
Topology Rapid Configuration. From the drop-down menu choose Bus and click OK. 2. Click the Select Models button in the Rapid Configuration dialog box. From the Model List drop-down menu choose ethcoax and click OK. 3. In the Rapid Configuration dialog box, set the following eight values and click OK.
4. To configure the coaxial bus, right-click on the horizontal link Select Advanced Edit Attributes from the menu: a. Click on the value of the model attribute Select Edit from the dropdown menu Choose the eth_coax_adv model. b. Assign the value 0.05 to the delay attribute (propagation delay in sec/m). c. Assign 5 to the thickness attribute. d. Click OK.
5. Now you have created the network. It should look like the illustration below. 6. Make sure to save your project.
6. Click OK to return back to the Project Editor. 7. Make sure to save your project. (D) CONFIGURE THE SIMULATION: To examine the network performance under different loads, you need to run the simulation several times by changing the load into the network. There is an easy way to do that. 1. Click on the Configure/Run Simulation button: 2. Make sure that the Common tab is chosen Assign 15 seconds to the Duration.
3. Click on the Object Attributes tab. 4. Click on the Add button. The Add Attribute dialog box should appear filled with the promoted attributes of all nodes in the network (if you do not see the attributes in the list, close the whole project and reopen it). You need to add the Interarrival Time attribute for all nodes. To do that: a. Click on the first attribute in the list (Office Network.node_0.Traffic Generation .) Click the Wildcard button Click on node_0 and choose the asterisk (*) from the drop-down menu Click OK. b. A new attribute is now generated containing the asterisk (the second one in the list), and you need to add it by clicking on the corresponding cell under the Add? column. c. The Add Attribute dialog box should look like the following. Click OK.
5. Now you should see the Office Network.*.Traffic Generation Parameter in the list of simulation object attributes. Click on that attribute to select it Click the Values button of the dialog box. 6. Add the following nine values. (Note: To add the first value, double-click on the first cell in the Value column Type exponential (2) into the textbox and hit enter. Repeat this for all nine values.)
7. Click OK. Now look at the upper-right corner of the Simulation Configuration dialog box and make sure that the Number of runs in set is 9.
8. For each simulation of the nine runs, we need the simulator to save a scalar value that represents the average load in the network and to save another scalar value that represents the average throughput of the network. To save these scalars we need to configure the simulator to save them in a file. Click on the Advanced tab in the Configure Simulation dialog box. 9. Assign <your initials>_Ethernet_Coax to the Scalar file text field.
with
OBSERVATION:
1. Select View Results (Advanced) from the Results menu. Now the Analysis Configuration tool is open. 2. Recall that we saved the average results in a scalar file. To load this file, select Load Output Scalar File from the File menu Select <your initials>_Ethernet-Coax from the pop-up menu. 3. Select Create Scalar Panel from the Panels menu Assign Traffic Source.Traffic Sent (packets/sec).average to Horizontal Assign Traffic Sink. Traffic Received (packets/sec).average to Vertical Click OK.
LAB INSTRUCTIONS FOR CARRYING OUT PRACTICAL ON STUDY OF RESOURCE RESERVATION PROTOCOL (RSVP) AS A PART OF THE INTEGRATED SERVICES APPROACH FOR PROVIDING QUALITY OF SERVICE (QOS) TO INDIVIDUAL APPLICATIONS OR FLOWS.
AIM:
To study the Resource Reservation Protocol (RSVP) as a part of the Integrated Services approach for providing Quality of Service (QoS) to individual applications or flows.
THEORY:
For many years, packet-switched networks have offered the promise of supporting multimedia applications, that is, those that combine audio, video, and data. Audio and video applications are examples of real-time applications. The best-effort model, in which the network tries to deliver your data but makes no promises and leaves the cleanup operation to the edges, is not sufficient for real-time applications. What we need is a new service modelone in which applications that need better assurances can request such service from the network. The network may then respond by providing an assurance that it will do better, or perhaps by saying that it cannot promise anything better at the moment. A network that can provide different levels of service is often said to support QoS. Two approaches have been developed to provide a range of QoS: Integrated Services and Differentiated Services. The Resource Reservation Protocol follows the Integrated Services approach, whereby QoS is provided to individual applications or flows. The Differentiated Services approach provides QoS to large classes of data or aggregated traffic. While connection-oriented networks have always needed some sort of setup protocol to establish the necessary virtual circuit state in the routers, connectionless networks like the Internet have had no such protocols. One of the key assumptions underlying RSVP is that it should not detract from the robustness that we find in the Internet. Therefore, RSVP, uses the idea of soft state in the routers. Soft statein contrast to the hard state found in connection-oriented networksdoes not need to be explicitly deleted when it is no longer needed. Instead, it times out after some fairly short period if it is not periodically refreshed.RSVP adopts the receiver-oriented approachthe receivers keep track of their own resource requirements, and they periodically send refresh messages to keep the soft state in place. In this experiment you will set up a network that carries real-time applications and that utilizes RSVP to provide QoS to one of these applications. You will study how RSVP contributes to the performance of the application that makes use of it.
4.
5. Click on WFQ and rename it to QoS_RSVP Click OK. 6. Make sure that you have only one scenario in your project named QoS_RSVP. The following figure shows one way to check for the available scenarios in the project.
3. Click on the Voice Called node to select it From the Edit menu, select Copy From the Edit menu, select Paste (alternatively, use the standard keyboard shortcuts, Ctrl-C and Ctrl-V). i. Locate the new node somewhere below the Voice Called node on the screen Connect the new node to the East Router using a 10BaseT link. ii. Right-click on the new node Edit Attributes. iii. Click on the ethernet_wkstn value of the model attribute Select Edit Select the ethernet_wkstn_adv model. iv. Rename it to Voice_RSVP Called Assign Voice_RSVP Called to its Client Address attribute. v. Click OK. 4. Copy and paste the Voice Caller node. i. Locate the new node somewhere below the Voice Caller node Connect the new node to the West Router using a 10BaseT link. ii. Right-click on the new node Edit Attributes. iii. Click on the ethernet_wkstn value of the model attribute Select Edit Select the ethernet_wkstn_adv model.
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iv. Rename it to Voice_RSVP Caller. v. Edit the Application: Destination Preferences attribute Open the Actual Name table by clicking in the value field Actual Name Assign Voice_RSVP Called to the attribute. vi. Click OK three times. 5. Rename the Queues node in the project to QoS. Your project should look like the following diagram. 6. Save your project.
The flow is defined by its required bandwidth and buffer size. Bandwidth is set to be the token bucket rate in the flow specification of the Path and Resv messages. The buffer size represents the amount of the application bursty data to be buffered. It specifies the token bucket size that will be set in the Path or Resv messages for the session. 1. Right-click on the QoS node Edit Attributes. i. Expand the RSVP Flow Specification hierarchy and its row 0 hierarchy Set Name to RSVP_Flow Assign 50,000 to Bandwidth (bytes/sec) attribute Assign 10,000 to the Size (bytes) attribute. ii. Expand the RSVP Profiles hierarchy and its row 0 hierarchy Set Profile Name to RSVP_Profile. iii. Click OK and then save your project.
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ii. Click on the PCM Quality Speech value (shown above) Select Edit Edit the value of the RSVP Parameters attribute Assign the following values (recall that we defined the RSVP_Flow in the QoS node) Click OK three times.
Note that the characteristics of the Outbound Flow are carried in the Path messages to be sent from sender to receiver, and the characteristics of the Inbound Flow parameters are carried in the Resv messages to be sent from the receiver to the sender.
2. From the Protocols menu, select RSVP Select Configure Interface Status Make the selections shown below in the configuration dialog box Click OK and then save your project.
The above process enables RSVP on all interfaces along the path between the two Voice parties that need to utilize RSVP.
iii. Expand the IP Host Parameters hierarchy Expand its Interface Information hierarchy Expand the Information hierarchy Assign WFQ Queuing Scheme attribute Assign ToS Based Queuing Profile attribute Assign RSVP Enabled RSVP Info attribute.
iv. Expand the RSVP Protocol Parameters hierarchy Expand the Interface Information hierarchy. (You should notice that the word Enabled is listed in the summary line. When you expand it, you will see that it is the value of RSVP Status. If Enabled is not listed, go back to the Configure the Interfaces steps.) Expand the hierarchy of the row of that interfaces Assign 75% to both the Maximum Reservable BW and Maximum Bandwidth Per Flow attributes as shown:
v. Click OK. 2. Right-click on the Voice_RSVP Called node Edit Attributes. i. Edit the Application: Supported Services attribute. The Application: Supported Services Table will popup In that table, replace the VoIP Application with VoIP_RSVP and click OK. ii. Expand the Application: RSVP Parameters hierarchy Expand its Voice hierarchy Enable the RSVP Status Expand the Profile List hierarchy Edit the value of the attribute of row 0 and write down RSVP_Profile. iii. Expand the IP Host Parameters hierarchy Expand its Interface Information hierarchy Expand the QoS Information hierarchy Assign WFQ to the Queuing Scheme attribute Assign ToS Based to the Queuing Profile attribute Assign RSVP Enabled to the RSVP Info attribute. iv. Expand the RSVP Protocol Parameters hierarchy Expand the Interface Information hierarchy. (You should notice that the RSVP Status of the interface that is connected to the router is Enabled. If not, go back to the Configure the Interfaces steps.) Expand the hierarchy of the row of that interface Assign 75% to both Maximum Reservable BW and Maximum Bandwidth Per Flow attributes. v. Click OK. 3. Right-click on the East Router node Edit Attributes. i. Click on the Ethernet4_slip8_gtwy value of the model attribute Select Edit Select the Ethernet4_slip8_gtwy_adv model. ii. Expand the RSVP Protocol Parameters hierarchy Expand the Interface Information hierarchy. (You should notice that the RSVP Status of two interfaces, which are connected to the West Router and the Voice_RSVP Called node, are Enabled. If not, go back to the Configure the Interfaces steps.) Expand the hierarchies of the rows of these two interfaces Assign 75% to both Maximum Reservable BW and Maximum Bandwidth Per Flow attributes. iii. Expand the IP Routing Parameters hierarchy Expand the Interface Information hierarchy Expand the hierarchies of the rows of the same two interfaces you configured in the previous step (step ii) Expand the QoS Information hierarchy for both Set Queuing Scheme to WFQ and
Profile
Queuing Profile to ToS Based for both. iv. Click OK. 4. Right-click on the West Router node Edit Attributes. i. Click on the Ethernet4_slip8_gtwy value of the model attribute Select Edit Select the Ethernet4_slip8_gtwy_adv model. ii. Expand the RSVP Protocol Parameters hierarchy Expand the Interface Information hierarchy. (You should notice that the RSVP Status of two interfaces, which are connected to the East Router and the Voice_RSVP Caller node, are Enabled. If not, go back to the Configure the Interfaces steps.) Expand the hierarchies of the rows of these two interfaces Assign 75% to both Maximum Reservable BW and Maximum Bandwidth Per Flow attributes. iii. Expand the IP Routing Parameters hierarchy Expand the Interface Information hierarchy Expand the hierarchies of the rows of the same two interfaces you configured in the previous step (step ii) Expand the QoS Information hierarchy for both Set Queuing Scheme to WFQ and Queuing Profile to ToS Based for both. iv. Click OK.
5. Expand the Voice Calling Party hierarchy and select the following statistics: Packet Delay Variation and Packet End-to-End Delay (sec). 6. Click OK.
on the Voice_ RSVP Called node and select Choose Individual Statistics from the pop-up menu.
2. Expand the RSVP hierarchy and select Number of Resv States. 3. Right-click on the Number of Resv States statistic Select Change Draw Style from the pop-up menu Choose bar chart. 4. Right-click on the Number of Resv States statistic Select Change Collection Mode from the pop-up menu Check the Advanced checkbox From the Capture mode drop-down menu, select all values Click OK. 5. Click OK.
Voice Caller Statistics: 1. Right-click on the Voice Caller node and select Choose Individual Statistics from the pop-up menu. 2. Expand the Voice Calling Party hierarchy and select the following Statistics: Packet Delay Variation and Packet End-to-End Delay (sec) 3. Click OK. Configure the Simulation 1. Click on and the Configure Simulation window should appear. 2. Make sure that the duration is set to 150 seconds. 3. Click on the Global Attributes tab and make sure that the following attribute is enabled: a. RSVP Sim Efficiency = Enabled. This decreases the simulation time and memory requirements by not sending refresh messages (i.e., Path and Resv refreshes). 4. Click OK and then save your project. Run the Simulation: 1. Click on and then click the Run button. Depending on the speed of your Processor, this may take several minutes to complete. 2. After the simulation completes, click Close. 3. Save your project.
for both the Voice Caller and Voice_RSVP Caller nodes. Choose Overlaid Statistics and time_average.
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3. Click Show to get the following graph. (Note: To zoom in on the graph, click and drag your mouse to draw a rectangle around the area interest and release the mouse button.)
4. Similarly, you can get the following graph that compares the Packet Delay Variation for both the Voice Caller and Voice_RSVP Caller nodes. (Note: Make sure to unselect the statistics you chose for the previous graph.)
5. Finally, prepare the graph that displays the number of Path and Resv states by selecting the following statistics. Make sure to select Stacked Statistics and As Is as shown.
6. Right-click on the resulting graph and choose Edit Panel Properties Change the assigned values to the Horizontal Min and Horizontal Max fields as shown (your graph might require a slightly different range):
7. Click OK. The resulting graph should resemble the one below.