The Analysis and Simulation of VoIP

ENSC 427 Communication Networks Spring 2013 Final Project The Analysis and Simulation of VoIP http://www.sfu.ca/~cjw11/427project.html Group #3 Demet Dilekci ddilekci@sfu.ca Conrad Wang cw11@sfu.ca Jiang Feng Xu jfxu@sfu.ca

Table of Contents Abstract... 2 Glossaries... 3 1. Introduction... 4 1.1 Objective... 4 1.2 Background... 4 Advantages... 4 Disadvantages... 4 1.3 Design Scenarios... 4 1.4 Parameters... 5 Jitter... 5 End-to-End Delay... 6 Packet Loss... 6 Mean Opinion Score (MOS)... 6 1.5 Parameter Standards... 6 2. Design Implementations... 7 2.1 LAN Configuration... 7 2.2 WAN Configuration... 8 2.3 WLAN Configuration... 9 2.4 WWAN Configuration... 9 3. Simulation Results and Discussions... 11 3.1 LAN Results... 11 3.2 WAN Results... 14 3.3 WLAN Results... 17 3.4 WWAN Results... 20 4. Conclusion... 25 References... 26 1 P a g e

Abstract Recently, a communication method called Voice over Internet Protocol (VoIP) has grown to be very popular, and it is gradually replacing Public Switched Telephone Network (PSTN) as the best choice for voice communication. VoIP is a protocol that allows people to make calls over the internet. It has many advantages over the traditional telephone line such as the cheap cost and the ability to perform group conversations. However, it also has some drawbacks such as jitter, delay/latency, packet loss, and voice compression. In this project, we will analyze and evaluate the VoIP performance in terms of these issues by simulating different LAN, WAN, WLAN, and WWAN scenarios in OPNET 16. 2 P a g e

Glossaries ITU International Telecommunication Union LAN Local Area Network MOS Mean Opinion Score PSTN Public Switched Telephone Network VoIP Voice over Internet Protocol WAN Wide Area Network WLAN Wireless Local Area Network WWAN Wireless Wide Area Network 3 P a g e

1. Introduction 1.1 Objective The objective of this project is to create Ethernet and wireless network scenarios for VoIP and compare the simulation results by using OPNET 16.From these results, we will determine whether VoIP is a suitable substitute over PSTN. 1.2Background VoIP is a method of transmitting voice signal over the internet instead of transmitting over the land lines. VoIP converts the analog signal of the voice into digital signal. Advantages VoIP has many advantages over the traditional PSTN. The most important advantage of VoIP is that is cheap to use. There is no charge for VoIP calls as long as you have an internet service provider. This also means that there is no additional charge for long distance calls, caller display, or any other features. Another big advantage of VoIP is its rich features. You are able to do conference calls, video calls, or even send files when having conversations. Disadvantages While VoIP has major advantages over PSTN, it also has some drawbacks. Firstly, it is internet dependent. If the internet is not available or when the service is poor, it ll largely affect the call quality. Also, since VoIP uses packet switching rather than circuit switching, the connection reliability is much lower and the drop rate is higher. Lastly, 911 call is unable to trace your location if you are using VoIP, making VoIP not an option in emergency situations. 1.3 Design Scenarios The following are the different Ethernet and wireless network scenarios that we will simulate so that we can understand the performance of VoIP in different situations. 4 P a g e

Ethernet Office (LAN) Long Distance (WAN) 2 client 20 client 2 client 20 client Figure 1.3.1: Ethernet Scenarios Wireless Office (WLAN) Long Distance (WWAN) 2 client 10 client 2 client 10 client Figure 1.3.2: Wireless Scenarios 1.4 Parameters In each scenario, there are 4 major parameters that we will focus on when reading from the simulation results. Jitter Jitter is the variance of time between each packet arrival. This occurs due to network congestion. The issue can be resolved by adding jitter buffers. This parameter is important in communications since call quality rely heavily on the amount of jitter. High jitter would lead to poor call quality because voice information would not be received in a timely fashion, and the information would not make sense. 5 P a g e

End-to-End Delay End-to-end delay is the difference in time between when the packet is sent and when the packet is received. This usually occurs due to network performance and the distance between two nodes. This parameter is important because we want to receive voice information in real-time. If there is too much delay, it would be difficult to have a conversation. Packet Loss Packet loss is the rate in which packets sent do not arrive at the receiving end. This is because VoIP uses packet switching, so there is no guarantee that packets arrive. This parameter is important because losing information in a call would make it extremely hard to communicate. Mean Opinion Score (MOS) MOS is the grading scheme for the quality of a VoIP call. It is graded by the user from a scale of 1 to 5, bad to excellent. The score is determined by several of factors such as jitter, end-to-end delay, and packet loss. 1.5 Parameter Standards According to the International Telecommunication Union (ITU) standards, the average and ideal quality values for the parameters are: Average Quality Ideal Quality Jitter < 60 ms < 20 ms End-to-end Delay < 150 ms < 50 ms Packet Loss Rate < 5% < 1% Table 1.5.1: ITU Standards for Jitter, ETE Delay, and Packet Loss Quality Scale Mean Opinion Score (MOS) Excellent 5 Good 4 Fair 3 Poor 2 Bad 1 Table 1.5.2: MOS Scale 6 P a g e

2. Design Implementations 2.1 LAN Configuration In the LAN configuration we use two topologies; a simple network including 2 workstations and a LAN topology including two star topologies that are connected by a Cisco 2500 router. Figure 2.1.1: OPNET VoIP over LAN Designs (2 and 20 client) Both network includes light e-mail, ftp and http traffic belong the VoIP traffic. VoIP table, Application and profile attribute definitions are given below. Figure 2.1.2: OPNET VoIP Application, Profiles and Voice Table definitions 7 P a g e

2.2 WAN Configuration Inside the subnet, we have the same configuration for the two clients. For the twenty clients, we place a star topologies consisting of ten clients in each side. Figure 2.2.1: OPNET VoIP over WAN Designs (2 client) Figure 2.2.2: OPNET VoIP over WLAN Designs (20 client) 8 P a g e

2.3 WLAN Configuration In the wireless configuration, we use the standard of IEEE 802.11g with 54 Mbps. Also, in the layout, we place the router in the center of clients. Figure 2.3.1: two clients in office For the ten clients, we basically increase the number of clients around router. Figure 2.3.2: Ten clients in office 2.4 WWAN Configuration For the long distance case, we choose the network which the two locations are relative far always from each other (Vancouver and Toronto). Since the router has limited effective range, we need to use Ethernet to connect two wireless routers in long distance. Also, we want to investigate the effect of the connection link; thus, we use different links which are PPP_DS1, PPP_DS3, 10 BaseT, and 100 BaseT. Inside the subnet, we have the same configuration for 9 P a g e

the two clients. For the ten clients, we place five clients in each side. Figure 2.4.1: long distance layout 10 P a g e

3. Simulation Results and Discussions 3.1 LAN Results The jitter as seen the steady part of the figure 3.1.1, is around zero. Close to the end of simulation time, there is a sudden increase (still in good level) in the jitter as well as other performance values. Since we focused on the steady part, we ignore it. Conversation pairs: Node 6 (V) & node 11(T) Node 6 (V) & Node 19 (T) Figure 3.1.1: Jitter for office For the MOS value, both 2 and 20 client topologies have same MOS values, around 3.7 which is close the good level 4. We can see that the increasing the number of client from 2 to 20 in LAN, doesn`t affect the MOS. 11 P a g e

Figure 3.1.2: MOS value office For the ETE Delay, both 2 clients and 20 client s case have 50 ms ETE which is in good level. Although it is expected that increasing number of the client may degrade Jitter, ETE Delay, and MOS, they didn t. Figure 3.1.3: ETE Delay office 12 P a g e

As a result we don`t expect to see packet loss in VOIP under these scenarios. Figure 3.1.4 show that there are the same send and receive packets, which meet our expectation. Figure 3.1.4: Voice packet sent and received office 13 P a g e

3.2 WAN Results As it can be seen from the figure 3.2.1, the jitter is close to zero for 2 clients VOIP. When number of client increases to 20, the jitter increases as expected. However the increase is slight, it is still in good level. Conversation pairs: Node 6 (V) & node 11 (T) Node 1 (V) & Node 2 (T) Figure 3.2.1: Jitter long distance For the ETE Delay, we can see from figure 3.3.2, since the number of client is larger in node 11 than in node 1 and 2, it has larger delay and is in good level. However, while node 6 in Vancouver has 70 ms ETE (even lower than 2 client topology), node 11 in Toronto has 83 ms ETE. Both nodes are conversation pair. Similarly, while node 11 has 3.56 MOS, node 6 has 3.64 MOS. 14 P a g e

Figure 3.2.2: ETE Delay Figure 3.2.3: MOS Value long distance 15 P a g e

For the packets loss, the sent and received traffic packets is same. Even though, we change the client number, there is no packet loss during the transmission. Figure 3.2.4: Voice packet sent and received long distance 16 P a g e

3.3 WLAN Results First, we look at the Jitter in office case. As we can see in figure 3.3.1, when we look at the steady part of the figure, we can state that Jitter is somehow around to the zero. However, we expect the time delay from the Jitter. Figure 3.3.1: Jitter for office For the MOS value, the first two lines correspond to the two clients case. We can see that the MOS value is close to each other. The lower two line correspond to the ten clients case. We can see that the MOS value is decreasing due to the increasing clients. In average, MOS value is equal to 3.688 for two clients and 3.66 for ten clients. Comparing the theoretical value we find, we can conclude that the MOS value fall into an acceptable range. 17 P a g e

Figure 3.3.2: MOS value office For the ETE Delay, the two clients case is around 61ms. In the ten clients case, we can see that the delay is increasing significantly. In average, ETE Delay is equal to 66ms for the ten clients case. Also, the incensement is reasonable since we increase the number of clients. Also, comparing to the theoretical ETE Delay value, the result falls into the range between 50ms to 150ms. 18 P a g e

Figure 3.3.3: ETE Delay office For the packet loss, we notice that there are the same send and receive packets from the figure show in below, which meet our expectation. Since the scenario is located in the office, the clients have limit distance among them. Thus, we expect no packet loss in the office which the figure conform it. 19 P a g e

Figure 3.3.4: Voice packet sent and received office 3.4 WWAN Results First, we use different links to connect the wireless routers. We can see from figure 3.4.1, the Jitter is around zero in average. When we look at the steady part of the figure, we can state that Jitter is somehow around to the zero. However, we expect the time delay from the Jitter since the scenario is long distance connection. 20 P a g e

Figure 3.4.1: Jitter long distance For the MOS value, since we change the link type of the connection, the MOS value change significantly. As we expect, the PPP_DS3 link will produce the best MOS value. Also, we can see the MOS value change due to the long distance communication. The list of the MOS value of the different type links. Name 10 BaseT 100 BaseT PPP_DS1 PPP_DS3 MOS Value 3.62 3.625 3.645 3.65 21 P a g e

Figure 3.4.2: MOS Value long distance For the ETE Delay, the effective of link types is significant. We can see from table 3.4.1, PPP_DS3 has the lowest Delay among the other types of the link. As expected, the 10 BaseT has the highest delay. Link names ETE Delay 10 BaseT 72ms 100 BaseT 71ms PPP_DS1 68ms PPP_DS3 67ms Table 3.4.1 Links and ETE Delay 22 P a g e

Figure 3.4.3: ETE Delay long distance For the packets loss, the sent and received traffic packets is same. Even though, we change different type links, there is no packet loss during the transmission. 23 P a g e

Figure 3.4.4: Voice packet sent and received long distance 24 P a g e

4. Conclusion VoIP over LAN gives good result as tested with: MOS (3.7) is close to good level (4), and jitter is around zero. ETE is 50ms which is in good level and there is no packet drop. Increasing number of the client doesn`t degrade the VoIP performance. VoIP over WAN also gives good results. MOS (3.56-3.64) is close the LAN value (3.7) and there are slight increase in ETE (50 ms in LAN to 77-83 ms in WAN) and jitter due to increasing number of client. ETE and jitter values are in good level. Also there is no packet drop. The only concern for the slight difference in ETE and MOS values of two conversation pair: node 6 in Vancouver has 70 ms ETE and node 11 in Toronto has 83 ms ETE. Similarly, while node 11 has 3.56 MOS, node 6 has 3.64 MOS. For wireless, we noticed that the outputs change significantly due to the link types and the distance. While we increase the distance of the connection, the MOS value is decrease and the ETE delay is increase. Also, different link types can produce the different output. Since we choose the better link, then the ETE Delay will decrease and the MOS value will be increase. For all the scenarios, we can conclude that the results are all within the acceptable range. Thus, VoIP is a good substitute to PSTN. In a future part of the project we plan to test VoIP under heavy traffic / more crowded network in LAN and WAN to further examine the performance of VoIP. 25 P a g e

References [1] J. Davidson, J. Peters, M. Bhatia, S. Kalidindi and S. Mukherjee, Voice over IP Fundamentals. Indianapolis: Cisco press, 2007 [2] Understanding Voice over IP Protocols", http://www.cisco.com/en/us/tech/tk652/tk701/tech_digests_list.html, 14 Feb 2013 [3] J. Middleton, "Top 11 Technologies of the Decade", IEEE Spectrum, vol 48, issue.1, pp. 34-37, January 2011 [4] J. Soares, S. Neves and C. Rodrigunes, "Past, Present and Future of IP Telephony", IEEE CTRQ Conference, pp. 19-24, 2008.12 [5] L. Chu, X. Lan, Y. Tan, "The Design and Simulation of Enterprise's VoIP Network", IEEE ICECE Conference, pp. 2653-2655, 2011 [6] E. Chi-Pong Chan, Performance Analysis of Voice Communications in a Private802.11 Network, Ensc 835: High-Performance Networks, 2003, pp. 8 [7] K. Alutaibi, Performance Analysis of VoIP over WiMAX and Wi-Fi Networks, Ensc 894: Communications Networks, 2012, pp. 3-4 26 P a g e