ENSC 427 COMMUNICATION NETWORKS SPRING Final Project Report
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1 ENSC 427 COMMUNICATION NETWORKS SPRING 2013 Final Project Report "End to End Delay Performance Evaluation VoIP in the LTE network Group #4 Dai, Hongxin Ishita, Farah Lo, Hao Hua sfu.ca sfu.ca sfu.ca
2 Table of Content List of Acronyms.3 Abstract Introduction LTE VoIP Description of Project Configuration and Design General Application Configuration General Application Profile Configuration LTE Configuration Mobility Configuration General Parameter FTP Parameter VoIP Parameter LTE Parameter Topology VoIP over LTE Network (scenario 1 and 2) VoIP Congested with FTP (scenario 3) Discussion and Analysis of Results Scenario Scenario Scenario Conclusion...22 Reference
3 List of Acronyms UL User Terminal FTP File Transfer Protocol TDD Time Division Duplex FDD Frequency Division Duplex VoIP Voice over Internet Protocol LTE Long Term Evolution EPC Evolved Packet Core QoS Quality of Service SGW Serving Gateway MME Mobility Management Entity DwPTs Downlink Pilot Time Slot UpTs Uplink Pilot Time Slot PLC Packet Loss Concealment 3
4 Abstract LTE is a fourth generation protocol, which is quite effective as an advanced source of modern technology. The topic of this project mainly focuses on the performance of the VoIP. Throughout the research we have tried to evaluate different scenarios that would give results particularly for VoIP over LTE protocol. The configurations and network topology set ups for this LTE based research were done on OPNET 16.0Modeler. We simulated different network scenarios and observed the obtained results for conducting the research. The obtained results from simulations were interpreted to reach conclusions for the project. 1.0 Introduction 1.1 LTE LTE also known as Long term Evolution is a fourth generation (4G) wireless broadband technology, that has evolved from the third generation (3G) technology. While conducting our project based on LTE, we have tried to study the issues regarding the traffic behavior of VoIP alone as well as with other noise/traffics within the LTE system. We would like to see further research and evaluation of this superior and highly advanced technology. Since it is still new to the current time, it still has flaws that need to be fixed in the future time period. Nonetheless, we have carried out our project while having prime focus on the VoIP over LTE. This highly advanced technology is enormously used over video streaming and voice over internet protocol. The architecture of LTE consists of SGW, as well as MME. Figure 1. LTE Architecture [1] 4
5 Due to the dependence of core network of LTE on IP protocol, the core network of LTE is known as EPC. EPC consists of SWG and MME. Serving Gateway is mainly responsible for routing and forwarding packets. MME is basically involved in managing the mobility of packets. The evolved network nodeb (enbs) are the source and destination nodes of the LTE networking system. Packet Data Network Gateway is a form of a network element of LTE architecture that provides QoS between the 3rd generation and non-third generation technologies. LTE consists of two different kinds of frame structures as well, such as: type 1 and type 2. They are applicable to FDD and TDD respectively. Type 1 frame structure consists of a frame, which basically has a time period of 10 ms, each frame is divided in to 20 slots. Each frame of LTE contains 10 frames. Figure 2. Type 1 LTE frame structure [2] The type 2 structure of LTE has two identically similar half frames, where each half frame has a time period of 5 ms. Two half frames consists of 10 sub frames all together. Figure 3. Type 2 LTE frame structure [3] Each sub frame consists of DwPTs, UpTs and guard period. A guard period is required for preventing a clash of transmission of packets between the uplink and downlink.guard period is only required for the transmission of data packets from the UpTs to DwPTs. This is because, uplink is automatically synchronized to determine the time period of each sub frame. 5
6 1.2 VoIP VoIP is also known as Voice over Internet Protocol. Figure 4. VoIP [4] VoIP is involved in delivering voice communication over the computer networks. It has the internet packet-switching capabilities. The biggest advantage of VoIP is that it is actually a lot cheaper than the traditional phone calls and it has high bandwidth efficiency. VoIP connects wirelessly to the Wi-Fi or ether net due to their connection with the IP network. The data-link and physical layers provide a QoS mechanism for ensuring a better service for VoIP throughout the network. 6
7 2.0 Description of the Project The project consists of 3 main scenarios, 1. VoIP with different codec over LTE network 2. VoIP over LTE network while the senders and receivers are moving at different speed 3. VoIP over LTE network with FTP background traffic 2.1. Configuration and Design General Application Configuration Figure 5. Application Configuration In the application configuration, the applications and technologies that are used for the project can be chosen from the table. In this project, we use the voice and FTP application while voice application configuration details are shown on the figure above and FTP application configuration is set to be heavy load. 7
8 2.1.2 General Application Profile Configuration Figure 6. Application Profile Configuration The application profile configuration is used to regulate the behaviour of the application. For voice application, we will be setting the start time, duration and repeatability of the voice call here. For FTP application, except for those options in voice application, we also set the file size of the file that we want to transmit by FTP here. 8
9 2.1.3 LTE Configuration Figure 7: LTE Configuration In the LTE configuration, we make the LTE network specification to be all the same in all of the proposed scenarios. 9
10 2.1.4 Mobility Configuration Figure 8: Mobile Configuration The mobility configuration node is used to control the behaviour of mobile nodes regarding to their trajectory or speed. In this project, we will apply the mobility configuration specification to mobile objects in some scenarios in order to vary the speed of the nodes. 10
11 3.0 General Parameters Different parameters were used while configuring the networks: 3.1 FTP parameters FTP parameter Value Command Mix 50% Inter Request Time 60s File Size 1Gb/5Gb Type of Service Best Effort Table 1. FTP Parameters 3.2 VoIP parameters VoIP parameter Encoder Scheme Interactive Voice Offset Time Start Time Repeatability Duration Value G.711/G.723/G.729 Voice(6) 60s 40s Every 10s End of simulation Table 2. VoIP parameters The VoIP calls are made at a fixed interval of 60 seconds and the first initial call over VoIP is made after the first 100 seconds of the simulation. Once the first call has been made, a completely new VoIP call is added after one second interval of the time period. These non-stop calls over a gap of one second are made until the simulation procedures terminate completely. 3.3 LTE Parameters LTE Parameter QoS class identifier (Voice) Qos class identifier (FTP) UL Guaranteed Bit Rate DL Guaranteed Bit Rate UL Maximum Bit Rate DL Maximum Bit Rate UL Bandwidth DL Bandwidth Value 1 GBR 6 (Non-GBR) 1Mbps 1Mbps 1Mbps 1Mbps 20MHz 20MHz Table 3. LTE parameters The table above lists all the LTE parameters used for the simulation process. The above table stated that the UL (User Terminal) requires a band width of 20 MHz, which implies that the 11
12 users are able to transmit and receive signals for this given value of frequency. The data links (both uplink and down link) are in units of Megabits per second, which are 1Mbps in this case as seen from the table 3. These data bit rates are a measure of bits that can be transmitted per unit of time. The theoretical values of the data bit rates are approximately 4 times than the practical data bit rates. 12
13 4.0 Topology 4.1 VoIP over LTE network (Scenario 1 and 2) Figure 9. VoIP Network Topology OPNET 16.0 is used as a network simulator for the Baseline Voice of Internet protocol. The 3 network configurations are discussed. We can see from the VoIP topology that the sources and destinations are connected via EPC. Each enb has three VoIP workstations as seen from the VoIP topology. In scenario one, the speed of mobile objects are chosen to be 0m/s, 10m/s, 20m/s. Case Bandwidth VoIP Traffic Workstation Speed (m/s) (Mhz) Load (%) Distance (Km) Table 4. Cases of simulation for scenario 1 13
14 In scenario two, the different voice codes are chosen to sample the source information. The speed of the mobile objects is equally set to be 10m/s. Case Bandwidth Encoder Workstation Speed (m/s) (MHz) Scheme Distance (km) 1 20 G G.729A G k Table 5. Case of simulation for scenario VoIP congested with FTP (Scenario 3) Figure 10: Network topology of VoIP congested with FTP The figure (11) shows the network topology of VoIP mixed/congested with FTP traffic. The background noise or traffic is mainly generated due to the FTP. The Guaranteed and Non- Guaranteed Bit Rates were used for transmitting and receiving traffic for voice and FTP respectively. Four different cases were used for the simulation of the mixed VoIP and FTP traffic. The cases for the simulation of the mixed VoIP and FTP are listed in the table (6) below: 14
15 Case Bandwidth (MHz) FTP File Size (Gb) Speed (m/s) Table 6. Case of simulation for scenario 3 15
16 5.0 Discussion and Analysis of Results In the project, we want to determine the QoS of the network by analyzing end-to-end delay and packet loss rate of the network. End-to-End delay is the time required for a packet to travel from source (sender) to the destination (receiver). The unit for end-to-end is measured in seconds. Theoretically the End-to-End delay is not expected to go over a time period of 150ms. The three types of end-to-end delays with a Voice over Internet protocol are: sender delay, network delay, receiver delay. The packet loss takes place due to the unsuccessful transmission of packets from the source to their respective destinations. The packet loss over Voice over Internet Protocol (VoIP) is results into jitter and unexpected gaps over the received signal. The received signals may get completely dismissed due to the severe and unexpectedly occurred packet loss. VoIP usually uses the technique called PLC to minimize the loss of the packet over the network. PLC that involves the reception of artificially generated packets before the loss of actual packets.though PLC increases the latency; it is pretty effective to reduce the loss of packets Scenario 1 Figure 11. End-to-End Delay of scenario 1 16
17 In figure 12, the green, red and blue lines represent the case when the speed of mobile objects is 20m/s, 10m/s and 0m/s. The figures shows that the end to end delay for the case when the speed is 20m/s is the largest among all the cases, the second largest is when the speed equals 10m/s, then the smallest one is the case when the speed is 0m/s. The result matches our expectation. Theoretically, the end to end delay threshold for VoIP service is 150ms. Our result shows that the delay is around half of the threshold. It simply means for the case when speed of the mobile objects is less than 20 m/s, the VoIP s quality is considered to be acceptable. Figure 12. Packet Loss of scenario 1 In figure 12, the lines represent the amount of packets which are sent from source or received at the destination. We can see the lines are all overlapped with each other. It means there is no significant packet loss during the transmission for all of the above cases. The reason of that is for the voice application, the number of the packets that we are sending is small, with the provided 17
18 20MHz bandwidth, the number of the packets that we can send is far greater than the number of packets that we use in VoIP. 5.2 Scenario 2 Figure 13. End to End Delay of scenario 2 There are 3 kinds of encoder scheme, G.711, G K, G.729A, are used in this scenario. The different between theses 3 encoder schemes is that they use different sample method. The table below provides the information of these 3 encoder schemes. Encoder Scheme Data Rate (Kbps) Delay (ms) G G K G.729A Table 7. Codec Details As the table shown, when the G.711 encoder scheme is used (blue), the VoIP service will have the minimum delay, furthermore, when G.723 (red) encoder scheme is used, the VoIP service will have the maximum delay. The result that we obtain matches the expectation we have before the simulation. 18
19 Figure 14. Packet Loss of scenario 2 As shown in the figure 13, we can see that the packet loss rate is very small for encoder G.711 and G.729. However, at around 4 min 30 sec, it is obvious that for the encoder G.723 (yellow and red), the packet loss occurs. We believe at this point, due to the low data rate and high delay characteristic of encoder G.729, the voice packets actually congest during the transmission. As a result, some packets are dropped during the transmission. However, for the other two encoders, due to their better overall performance, the packets do not congest during transmission. 19
20 5.3 Scenario 3 Figure 15. End to End Delay of scenario 3 From figure 14, the blue, red, green and light blue lines correspond to the case of fixed object with 1GB FTP background traffic, fixed object with 5GB FTP background traffic, 20m/s mobile object with 1GB FTP background traffic and 20m/s mobile object with 5FB FTP background traffic. The green and light blue lines are overlapped. The result of the figure shows there is no significant difference when we increase the size of the file that we send. However, when the speed of the mobile objects increases, the delay increases as well. 20
21 Figure 16. Packet Loss of scenario 3 From figure 15, the blue, red, green and light blue lines indicate the packets that we receive at the destination. The packets that we send from source are same all the time. The figure shows the packet loss rate is proportional to both size of background traffic file and the speed of the mobile objects. *While we try to improve our implementation of the network for the last scenario, during the simulation, the simulation aborts and there is an error message saying virtual memory is full. As a result, we are unable to improve the result of this scenario. 21
22 6.0 Conclusion In the project, we conduct the analysis of QoS of VoIP over LTE network by studying the end to end delay and packet loss rate of the network in many different scenarios. The simulation is done by OPNET The following scenario has been analyzed: VoIP with different encoder scheme over LTE network, VoIP over LTE network while mobile objects are moving at different speed and VoIP over LTE network with FTP background traffic. Based on the simulation, we have found that in VoIP over LTE network case, the speed of the sender or receiver would affect the end to end delay performance of the application. The speed of the sender or receiver is proportional to the end to end. The speed does not actually affect packet loss rate. The future work could be done by increasing the speed of the mobile objects that we are examining such as 50 m/s, despite the fact that 50 m/s is not very practical in the real world nowadays. The second thing that we found is G.711 is so far the most appropriate encoder scheme for VoIP application. It gives the minimum end to end delay and packet loss. 22
23 References [1] LTE Architecture Diagram Gallery.[online]Available: [2] Type 1 LTE Sub frame. [Online]. Available: &bav=on.2,or.r_qf.&wrapid=tlif &um=1&ie=utf- 8&tbm=isch&source=og&sa=N&tab=wi&ei=iA5uUYC7MYTtiwLg0oGwCA [3] Type 2 LTE Sub frame. [Online]. Available: &bav=on.2,or.r_qf.&bvm=bv ,d.cge&wrapid=tlif &um=1&ie=utf -8&tbm=isch&source=og&sa=N&tab=wi&ei=xQ9uUcefBoeXiQK-94CABw [4] VoIP Network. [Online] Available: 2,or.r_qf.&wrapid=tlif &um=1&ie=UTF- 8&tbm=isch&source=og&sa=N&tab=wi&ei=chNuUbDLFIiviQK2pYC4Bw [5]3GPP, Third Generation Partnership Project (3GPP). [Online]. Available: [6]3GPP, Release 8 V0.0.3, Overview of 3GPP Release 8: Summery of all Release 8 Fea- tures, November [7]OPNET. Opnet modeler [Online]. Available: [8]Voice over IP. [Online]. Available: [9]Packet Loss [Online]. Available: [10] Technical White Paper: Long Term Evolution (LTE): A Technical Overview, Motorola, Inc [11] LTE Frame and Subframe Structure [Online] Available: [12] E. Masum, J. Babu, "End-to-End Delay Performance Evaluation for VoIP in the LTE network," Blekinge Institute of Technology,
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