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QoS Performance Evaluation of Voice over LTE Network

Ahmed J Jameel1* and Maryam M Shafiei2

1Department of Telecommunication Engineering, Ahlia University, Manama, Bahrain

2Department of Information Technology, Ahlia University, Manama, Bahrain

*Corresponding Author:
Ahmed J Jameel
Department of Telecommunication Engineering
Ahlia University, Manama, Bahrain
Tel: +973 36833172
E-mail: [email protected]

Received date: March 01, 2017; Accepyed date: March 16, 2017; Published date: March 24, 2017

Citation: Jameel AJ, Shafiei MM (2017) QoS Performance Evaluation of Voice over LTE Network. J Electr Electron Syst 6:216. doi: 10.4172/2332-0796.1000216

Copyright: © 2017 Jameel AJ, et al. This is an open-access article distributed under the terms of the Creative Commons Attribution License, which permits unrestricted use, distribution, and reproduction in any medium, provided the original author and source are credited.

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This paper describes the QoS performance evaluation of voice over LTE network using OMNeT++; an open-source system-level simulator and SimuLTE. OMNet++ is a well-known, widely-used modular simulation framework, which offers a high degree of experiment support. As a result, it can be integrated with all the network oriented modules such as INET. We describe the voice over LTE, and show performance evaluation results obtained using the simulator.


OMNeT++; INET; VoLTE; LTE; Voice over LTE simulation


In the past few decades, the mobile communication industries have evolved very fast to shift between each generation ranging from the 1G to LTE. This evolution from 1G to 4G was not as easy as it took a lot of work to make 4G technology the fastest network rollout. Long Term Evolution (LTE) was designed for the data transfer and also as a packet switched all-IP system. It does not contain any circuit switched domain for the purpose of providing with the regular voice and SMS services. The increase of the data traffic raised the issue of mobile broadband services by the consumers. The latest of these developments is the voice over LTE (VoLTE) it is devised scheme for standardized system between the mobile operators to carry out voice over Long Term Evolution (LTE) technology by replacing voice over the old technologies. The idea of the voice over LTE based on simply adapt to a completely new infrastructure based on internet protocol (IP) to replace the old legacy (2G-3G). The VoLTE specifications are based on air-interface, which is based on orthogonal frequency division multiplexing (OFDM) [1].

The new technology VoLTE can provide a combined system for transfer the voice traffic over the long term evolution air network access and employ the voice-over-IP (voice over internet protocol) technology, which is based on the (IP)- multimedia and IMS sub system to provide an appropriate service and video calling. The setup protocol for connection control is the session initiated protocol (SIP) which built to work with generic open IP network. The LTE is acclimatized with the current networks (3GPP, GSM, WCDMA, HSPA) and support for full forward and backward compatibility, until the LTE network voice service is fully implemented so the voice calls will automatic shift and full back to the best old bearer available (2G and 3G) [2].

This paper is organized as follows: section 2 presents the LTE network architecture, in section 3, the simulation results are presented in section 4, and section 5 concludes the paper.

LTE Network Architecture

The high level of the LTE network architecture as shown in Figure 1 is mainly composed of three main components:


Figure 1: LTE Network architecture.

UE (the User Equipment)

This component mainly consists of few of the functionalities of Mobile Terminal (MT) that is held responsible for all kinds of functioning of the call. On the other hand, the Terminal Equipment, which is also considered as one of the major devices of UE serves the function of data streaming and Universal Subscriber Identity Module (USIM). The USIM stores the network identification and user information. In this simulation of the LTE network, the User Equipment such as mobile, tablet, laptop etc., has been used [1].


The Evolved UTMS Terrestrial Radio Access Network used to handle the radio communication between the user equipment (UE) and the EPC. E-UTRAN composed of one or more base station called eNB or eNodeB. One of the parts of E-UTRAN is also termed as The Radio Access Network (RAN). The eNB or eNodeB serves the function of providing E-UTRA user plane and also it controls the plane protocol terminators along the user equipment [3].

EPC (Evolved Packet Core)

Evolved Packet Core is composed of two main elements: The Service Gateway (S-GW) which allows the user to communicate with other users of LTE network and PDN Gateway (P-GW) which is responsible to provide the connectivity between UE and external network like Internet. It serves the function of controlling the network access, management of mobility, and the other functions of network management. The Home Subscriber Server (HSS) present in the EPC stores all the information related with the subscriber. The entity of management of mobility controls the release and set-up of connections existing between the packet data network and user. It also accomplishes its activity through the registration of UE authentication location and using valuable information from the HSS. The Packet Data Gateway (P-GW) does the function of GGSN and SGSN, which also signifies the connectivity to the IP network. This system is assigned with the varied tasks of assignment of IP address, DHCP functions, user authentication, Quality of Service (QoS), charging data creation Deep Packet Inspection (DPI) [3].

Simulator Overview

Simulation software accomplishes a major role in the analysis of complex automation system and non-linear control system. Few of the software of computer that are designed for the dynamic system simulation at higher level than that of programming languages can be named as simulation languages, simulation software, simulation system, simulation environment and the simulators. Basically, simulation is explained as a particular method which is used for the solving of a problem in the dynamical systems, and which also finds out the model of the system rather than the real system. Simulation process follows few of the steps in sequence, which can be listed as formulation of problem, collection of data, mathematical modelling, identification of the model, and experiments with the model, representation of the result and interpretation of the result. Simulation software is usually used for designing, studying and analyzing the network communications. There are various software’s available in the market that can serve this purpose. Most of the simulation software’s are commercial but some of them are free for non-commercial use such as OMNET++.

OMNeT framework

OMNET++ is and extensible open-source library and framework primarily used to simulate networks. It can be used in various problem domains such as modelling of wired and wireless communications networks, evaluating performance aspects of complex networks. OMNET++ is widely used by academic institutions and educational environments for teaching purpose. It’s also used by students and researchers to study and analyze the performance of the networks.

The basic building block of OMNeT++ is modules, either simple module or compound module. These modules communicate through messages that are sent and received through connection linking the gates of the modules. OMNeT++ facilitates the user to keep the implementation, description and parameter values of the model separate. C++ is used as the coding of the implementation. The files written in Network Description (NED) language is used for expressing the description. Theses NEDs also allow for writing of the parametric topologies. The major reasons for selecting OMNeT++ as the major tool for the simulation is that it is one of the most mature, stable and enriched with features framework [4].

INET framework

OMNET++ has some of external extensions that can be used to design and simulate the wireless network such as INET Framework. The INET framework is an open-source model that should be installed on top of OMNET++. In addition to the wireless network, it can be used to simulate wired and mobile networks. It contains IPv4, IPv6, TCP, SCTP, UDP protocol implementations and some of the other application models. As that of OMNeT++, INET framework also uses the similar modules that communicate through the passing of message [5].


SimuLTE for OMNeT++ can be used to analyze and evaluate the performance of LTE and LTE Advanced networks. It is an open source project developed by group of researchers to evaluate the complex network environments. It should be installed on top of OMNET++ and INET Framework. It simulates the data plane of the LTE Radio Access Network and Evolved Packet Core. SimuLTE has a special feature that it contains around 40000 lines of codes that helps in the extra functionalities such as the applications, mobility, event queues, ID/UDP, and so on. However, in this particular study of the voice over LTE, this distinguished feature of SimuLTE has been extracted from the OMNeT++ and INET frameworks. [5].

Simulation Results

This section describes the implemented simulation topology in OMNET++, GUI and explains the simulation parameters used in the experiment.

Quality of Service (QoS) criteria

The performance of Voice over LTE can be measured with the help of various criteria. In this experiment the major focus will lie on the following four of the major criteria.

Mean Opinion Score (MOS): MOS is the grading system that is used for the measuring of the quality of a voice call. It is usually graded by the user with the scale of 1 to 5, which means bad to excellent. This particular score is determined by few of the factors such as end to end delay, jitter and packet loss. One of the empirical formula that can be used for the calculation of MOS score from the packet loss in terms of percentage in milliseconds is as follows [6].

MOS=ln( loss )-0.1ln( size ) (1)

The following Table 1 shows the standard and the ideal quality values for the Mean Opinion Score (MOS).

MOS Quality
5 Excellent
4 Good
3 Fair
2 Poor
1 Bad

Table 1: MOS standard.

End to end delay: End to End Delay is the time taken for a voice packet to be transmitted from the source UE to the destination UE across the LTE network. In simple words, it can be explained as the difference in the time between the sending and receiving of the packet. It basically takes place due to the performance of the network and the distance that exists between two of the nodes. This parameter is crucial so as to receive more information on the voice of a real time. There would be difficulty in having the effective communication in case of too much delay.

The following Table 2 shows the average and the ideal quality values for the VoLTE End to End Delay.

End to end delay Quality
<50 ms Ideal
<150 ms Average

Table 2: End to end delay standard.

Packet loss: Packet Loss can be defined as the number of the transmitted packets that are failed to reach its destination. It can also be described as the particular rate in which the packets that are being sent do not reach at the receiving end. The real time communications are based on the UD protocols. This protocol is usually without any connections and it cannot be send again if the packet is lost. The loss of the packages can also take place by removing all those packets that do not arrive to the end of the receiver on time. It becomes problematic whenever the loss of packet takes place in a bulk. The highest rate of packet loss so the voice can be heard with enough quality must be 1%.

The following Table 3 shows the average and the ideal quality values for the Packet Loss during Voice over LTE session.

Packet loss rate Quality
<1% Ideal
<5% Average

Table 3: Packet loss rate standard.

Jitter: Jitter is the variation in the latency of the voice packets sent from the source to the destination. This basically occurs due to the congestion in the network. These similar cases can be solved with the addition of jitters buffers. This is an important parameter to be considered while measuring the quality of service since the high jitter can lead to poor quality of voice. The high jitter usually leads to the weaker quality of call as the information of the voice will not be received within the timely manner and thus, the information will not make any sense. In the technical terms, jitter is the measure of the variability of the latency over the time and also across the network [7].

The jitter that exists between the starting and final point of the communication must always be less than 100 ms. If the value of the jitter becomes smaller than 100 ms, it can be adjusted with the addition of jitter buffers [7]. The following Table 4 shows the average and the ideal quality values for the Jitter:

Jitter Quality
<20 ms Ideal
<50 ms Average

Table 4: Jitter standard.

Simulation configuration

This section presents all the general parameters used in the conduction of the simulation.

• Ethernet Link Data Rate: 10 Mbps

• Simulation Time: 20 sec

• Packet Size: 40 byte

• Queue Size: 1 MB

Voice Over LTE Scenarios

Scenario (1 and 2) Voice over LTE network: OMNET++ architecture for the first and second scenarios is illustrated in Figure 2. The high level VoLTE network is composed by the following elements:

• Two User Equipment Support Voice Over LTE

• Two eNodeB

• Four Routers

• Two S-GW

• One P-GW

• One Internet Host


Figure 2: OMNeT++ LTE Network topology (Scenarios 1 and 2).

Scenario (1 and 2) Voice over LTE network: The third scenario of the VoLTE network (Figure 3) is composed with the help of below mentioned elements:

• Six User Equipment

• Two eNodeB

• Four Routers

• Two S-GW

• One P-GW

• One Internet Host.


Figure 3: OMNET++ LTE Network topology (scenario 3).

Simulation analysis and result

This section presents the simulation analysis and result for the conducted experiment. The quality of service can be measured by several of the factors. In this experiment, the quality of the network for each scenario has been compared in terms of MOS, End to End Delay, Packet Loss Rate and Jitter.

Scenario 1: This scenario has been implemented to conduct an evaluation analysis of the performance of VoLTE between two UEs. The speed of the sender and receiver of the voice is 0 m/s. The following Figures have been obtained after running the simulation of the first scenario.

MOS: The MOS of the first scenario stayed above 4 during the simulation which falls under the category ranging between the scale of good and excellent. The average of MOS we obtained is 4.36, which is the normal value of any VoLTE service.

The following Figure 4 shows the MOS obtained after running the simulation of Scenario 1


Figure 4: Scenario 1 MOS.

End to end delay: As we can see in Figure 5, we have delay for about 8 sec between the period 7 s-11 s. The average end to end delay is 1.77 ms which meets the standard since it’s below 50 ms.


Figure 5: Scenario 1 end to end delay.

Packet loss rate: According to the explanation and description of the packet loss, the average value obtained from the simulation is 0.21%. This rate is very small as compared to that of the ideal value (1%) (Figure 6).


Figure 6: Scenario 1 packet loss rate.

Jitter: The jitter as seen in Figure 7 remained static and didn’t change over the time. It stayed at the same value (8 ms) till the end of the call. Since the result is less than 20 ms this means that the quality of voice in this scenario was excellent.


Figure 7: Scenario 1 jitter.

Scenario 2: In this scenario, we studied the evaluation of the performance of VoLTE between the sender (UE1) and receiver (UE2). The simulation was conducted for about 20 s and the speed of the UEs during the voice conversation was 100 km/h (28 m/s).

The following Figures are obtained after running the simulation of the second scenario:

MOS: As the following Figure 8 shows, the MOS for the second scenarios is varying over the time. It started with 4.4, and then dropped to 1.7, then again increased to 4.4 and finally ended up with 1.7. The average value is 3.35 and based on the standard rating, it can be determined that the quality of the voice is ranging between fair and good.


Figure 8: Scenario 2 MOS.

End to end delay: As we can see in the Figure 9, the delay started with 0 ms then after 4 seconds from the beginning of the conversation it reached 8 ms then again it decreased to 0 ms. The average delay we got in this term is 2.6 ms, which is acceptable since it is less than 50 ms.


Figure 9: Scenario 2 end to end delay.

Packet loss rate: The average percentage of the packets loss for scenario 2 is 7.86%. This value is more than the ideal (1%) and the average (5%) percentage of the acceptable loss in VoLTE service. This variance in the percentage may affect the quality of the voice between the sender and the receiver.

From this result we can conclude that the speed of UEs during the VoLTE session possess the capacity of directly affecting the performance of the call (Figure 10).


Figure 10: Scenario 2 packet loss rate.

Jitter: The following figure shows the jitter result for the second scenario. As we can see from the chart, the jitter started to increase after 4 seconds from the start of the call. It increased from 8 ms to 23 ms and then dropped to 3 ms. The average jitter we got is 7.7 ms, which is acceptable as per the standard grading.

From the result we can conclude that making VoLTE call while driving may affect the quality of voice depending on the speed of the UEs (Figure 11).


Figure 11: Scenario 2 jitter.

Scenario 3: In this scenario, the sender (UE1) is calling the receiver (UE2) via VoLTE service while other four UEs are downloading a video of size 200 mb from the server. The other four UEs are connected to the same eNodeB as UE1.

MOS: The MOS of scenario 3 does not have a big difference when it is compared to the first Scenario (Figure 12). The MOS average value we obtained after conducting the simulation is 4.192, which can be considered to be close to the value of scenario 1 (4.3619).


Figure 12: Scenario 3 MOS.

End to end delay: The average delay value obtained from the simulation of scenario 3 is 2.5 ms which is much less than the ideal value (50 ms) and the average value (150 ms). This means that having a congested eNodeB while making a call in LTE network should not lead to a big delay in the packets sent (Figure 13).


Figure 13: Scenario 3 end to end delay.

Packet loss rate: The line chart in Figure 14 shows the packet loss during the simulation of scenario 3. As we can see the average percentage of the packet loss are less than 1% of the total packets sent which is acceptable (Figure 14).


Figure 14: Scenario 3 packet loss.

Jitter: The below mentioned diagram represent the jitter result from the third scenario. It can be seen that the jitter started to increase after 3 sec from the start of the call and again decreased to 1 ms after 17 sec. By 17 sec, it again increased to 7 ms and by 20 sec, it again declined to 1 ms Figure 15.


Figure 15: Scenario 3 jitter.

Scenarios result side by side: In this section, a brief comparison between the three scenarios based on the average QoS parameters values has been introduced.

The following Table 5 and charts compare the performance of the three conducted scenarios.

Scenarios QoS      
No. MOS End to end delay Packet loss Jitter
Scenario 1 4.3619 1.7 ms 0.0021 8 ms
Scenario 2 3.3489 2.6 ms 0.0786 7.7 ms
Scenario 3 4.1942 2.5 ms 0.0096 2.8 ms

Table 5: Scenarios comparison.

MOS: In the following diagram, the comparison of the voice call over LTE has been compared by keeping into consideration the three of the scenarios. Three of the different scenarios are displaying the irregular frequency of the VoLTE MOS vector.

End to end delay: The average delay value of the three scenarios has been compared in the above diagram. It can be observed that the first scenario is increasing at 7 sec and then again declining to o ms at 11 sec. The second scenario shows that at 4 sec, it is increasing and again it declines at slow rate. The final third scenario is showing a fluctuating rate with both increase and decline (Figures 16 and 17).


Figure 16: Scenarios MOS.


Figure 17: Scenarios end to end delay.

Packet loss: In the above diagram, the comparison of the packet loss during the simulation of three of the scenarios is demonstrated. The diagram shows a fluctuating rate in the different scenarios. Thus, if there is a fluctuation in the rate of packet loss, it might have adequate effect on the quality of voice between the sender and receiver [8,9].

Jitter: Similar to that of the above diagrams, in this, the comparison of the jitter results of all the three scenarios has been mentioned. The second scenario’s jitter result show that it is increasing from 4 sec and again at 20 sec, it declined. In the first scenario, the jitter result is seen to be stable throughout the call. Finally, the third scenario declines initially and then rises to certain point and again declines drastically (Figures 18 and 19).


Figure 18: Scenarios packet loss.


Figure 19: Scenarios jitter.


In this paper, performance analysis of Voice over LTE network is presented by studying the quality of service based on four of the major factors such as MOS, End to End Delay, Packet Loss Rate and Jitter. The simulation is designed and implemented with major simulation tools of OMNeT++ 4.6, INET Framework 2.6 and SimuLTE. Based on the simulated scenarios, we found that the speed of the sender and Receivers (UEs) are the crucial motivators that possess the capacity of seriously affecting the quality of the call. Once, the speed of the UEs is changed from 0 to 100 km/s, the average value of MOS has been dropped from 4.3619 to 3.3489.

The jitter and the packet loss percentage were also affected by the speed of UEs. Therefore, taking into consideration, all these facts and other measures, the simulation module of the LTE network with the help of OMNeT++, INET Framework and SimuLTE has been successfully done for this particular study. Apart from the above mentioned information, this particular section explains the operation of the LTE network under the variety of scenarios. Each of the scenarios explains the performance of data and voice under the different configurations. In all the scenarios, the description of the performance of voice has been explained according to the increase in the times of the general response due to the increase in the demand for traffic along with maximum bit rate from the different users and the maximum throughput.


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