International Journal of Advance Foundation and Research in Computer (IJAFRC)

Volume 2, Issue 5, May - 2015. ISSN 2348 4853

Performance Analysis Of VOIP Codecs With QoS

Parameters
Priyanka Grover1, Meenakshi Chawla2
Dept. of Computer Engineering,TIT&S,Bhiwani
parulgrover293@gmail.com, meenakshi.4441@gmail.com
Dept. of Computer Engineering,TIT&S,Bhiwani

ABSTRACT

This paper investigates the performance of popular VoIP codecs in the terms of some QoS related
parameters. This work is done over an IP network based on SIP architecture using RTP as a
transport protocol. VoIP is the service growing rapidly in the world of telecommunication. In
recent years VoIP(voice over internet protocol) is one of the most modern and interesting
technology. Opnet simulation tool is used here to analyze the performance of VoIP codecs. The
vital parameters to measure the quality of service are delay, jitter and throughput. The simulation
results give the best choice of codec for transporting the voice over IP backbone.

Index Terms: VoIP, G.711, G.723, G.729,SIP, QoS

I. INTRODUCTION

The transmission of real-time audio and video applications over the Internet is a challenging task. Of the
various real-time applications, Voice over IP (VoIP) has gained importance over the past few years owing
to its low cost and eases of interfacing between data and voice traffic.[9] VoIP have become the
expectation of next generation. It can provide unique and flexible networks as compared to the old
networks in which there was a separate network for data and for telephony. But now a single connection
can be used by a customer for data and voice communication which are transported by a service provider
using a single IP- based network.[11] To transmit voice over IP through any network, the voice is first
encoded at the transmitter side, and then transmitted to the receiver via communication channel. At
receiver side, the encoded signal is then decoded. Various VoIP codecs are used i.e G.711, G.729, G.723
which operate at different bit rates. VoIP is increasingly becoming an alternative to the traditional
PSTN(public switched telephone network) to send voice. The increase in the use of the real time services
such as VoIP has increased the demand for QoS. The QoS can be measured in the terms of throughput,
jitter, delay etc. In this paper, the performance of various VoIP codecs is analyzed in the terms of some
QoS related parameters. Based on the simulation results, we suggest a suitable codec of VoIP.

Rest of the paper is organized as follows. Section II gives an idea of VoIP technology. Section III describes
the methodology. Section IV presents the simulation results and discussion. Finally we conclude the
paper in section V.

II. VoIP TECHNOLOGY

1. VoIP Transport system

VoIP(voice over internet protocol) refers to a way to carry voice calls over a data network IP. VoIP
converts the analog voice signal from a telephone or computer into digital packetized signal that can be
transmitted over the internet. [7]This process is called encoding and the reverse of this process is called
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 International Journal of Advance Foundation and Research in Computer (IJAFRC)
Volume 2, Issue 5, May - 2015. ISSN 2348 4853

decoding. Both the processes are done by voice codecs. One of the main advantages of VoIP is its ability to
reduce cost because the calls go through the data network rather than the network operators telephony.
[10]Various signaling protocols are used by VoIP to deliver phone data over networks. SIP and H.323 can
be regarded as the protocols for VoIP services. [2]These protocols are used to set up, control and
terminate a VoIP call session. Once a session is established, voice stream is transported using transport
protocols, namely Real-time protocol(RTP) and User datagram protocol(UDP) to carry voice stream over
IP.[11]
2. VoIP Codecs
A VoIP codec is an algorithm used to encode and decode the voice stream. Different codecs uses different
algorithms to compress and decompress the voice stream. The codecs differ by the modulation and
demodulation technique such as Pulse code modulation(PCM), Algebraic code-excited linear
prediction(CELP) and Conjugate structure algebraic code excited linear prediction(CS-ACSELP).[9] Voice
over IP performance is based partially on the audio codecs which is used. The audio codecs are divided
into three types which are; Parametric codecs, Waveform codecs and Hybrid codecs. [5]ITU-T
Recommends a G.711 codec which is a non-uniform Pulse code modulation. The G.729 codec uses
Conjugate-Structure Algebraic-Code-Excited Linear-Prediction (CS-ACELP) . It provides voice calls at low
bit rate (i.e. 8 Kbit/s) according to ITU-T Recommendation. The G.729 consumes a high number of CPU
resources which limits the number of VoIP calls. G.723.1 is an audio codec for voice that compresses
voice audio in 30ms frames. There are two bit rates at which G.723.1 can operate: 6.3kbps (using 24byte
frames) and 5.3kbps (using 20byte frames) with Algebraic Code Excited Linear Prediction (ACELP)
algorithm. G.723.1 is mostly used in VoIP applications due to its low bandwidth requirement. The
International Telecommunication Union (ITU) has developed several standard codecs such as
G.711,G.718,G.719,G.720,G.723, and G.729. [3]Table 1 shows the characteristics of the G.711,G.723 and
G.729.
Table 1. VoIP Codec characteristics

IUT-T Algorithm Codec Bit Packet IP

Codec Delay Rate Per Packet
(ms) (kbps) second Size
(bytes)
G.711 PCM 0.375 64 100 120

G.729 ACELP 35 8 100 50

G.723 CS-ACELP 97.5 5.3 33 60

Table 1 shows that G.711 is a high bit rate codec, operating at a bit rate of 64 Kb/s whereas G.723.1 and
G.729A operate at low bit rates of 5.3 and 8 Kb/s respectively. This explained by the fact that G.711 uses
the basic Pulse Code Modulation (PCM) algorithm. On the other hand, complex algorithms are used by
G.723.1 and G.729A for compressing the bit rate resulting in encoding delays higher than for the basic
PCM algorithm. There is no license fee to use these codecs. G.711 works best in local networks where we
have a lot of available bandwidth. Each VoIP packet includes the headers at the various protocol layers
such as RTP, User Datagram Protocol (UDP), IP, 802.11 and the payload comprising the encoded speech
for certain duration depends on the codec deployed.[1] Researchers have further made significant
97 | 2015, IJAFRC All Rights Reserved www.ijafrc.org
 International Journal of Advance Foundation and Research in Computer (IJAFRC)
Volume 2, Issue 5, May - 2015. ISSN 2348 4853

contributions by modifying the flow and structure of various voice codecs to enhance the performance
efficiently keeping in view the standards and requirements of specific speech quality.[6]

3. QoS Parameters for VoIP

Quality of Service (QoS) is what determines if a technology can successfully deliver high value services
such as voice and video.[2] QoS is referred as the ability to control the mixture of bandwidth, delay, jitter,
and packet loss in a network in order to deliver a network service.[3] Ensuring high voice call quality
over the best effort IP network is the key challenge in delivering VoIP traffic. QoS for VoIP is defined
using different parameters, with the common ones being end-to-end delay, jitter and packet loss etc. The
BE class which is used for data stream with no support for delay and throughput. In this paper, the
quality of services for VoIP is measured in terms of end to end delay, jitter and packet loss. Delay or
latency is defined as the time required for a frame or a packet to travel from the source to its final
destination. The main source of delay is categorized into: Transmission Delay and destination processing
delay, capacity calculation ineffective or insufficient, technological constraints, reordering packets
queuing delay etc. [10] The variation of latency is jitter. An absolute value of delay difference between
selected packets to arrive at destination is called as jitter. It is not guaranteed that all the packets will
follow the same path and encounter the same routes to reach the destination over the network, and
added with the congestion in the network usually resulting in packets arrival out of order and with
varying delays. No jitter means a network with constant latency and no modification. The amount of data
that can actually be transmitted over the communication channel is called throughput. It is used to
estimate the efficiency of network. The ratio between the quantity of information and the sum of user
data, control data and retransmitted data if error is concluded as throughput of a network.

4. SIP(Session Initiation Protocol)

VoIP community uses SIP Protocol for signaling, registeration, admission control, transport, encoding,
setup control, initiate a session and facilitate real-time interactivity. It is an RFC(Request for comment)
standard from the internet engineering task force(IETF), responsible for the administering and
developing protocols that define the internet and developing protocols that define the internet.[8] SIP
translates the user name to the current network address, manages the call admission dropping, or
transferring mechanism, allows for changing the features of a session, etc. In the context of SIP, a user is
usually identified by an email- like address such as cleint@sip.ma, where client is a user name or phone
number, and sip.ma is a domain name or numerical address of the user. Some important parts of SIP are
User Agent (UA), Proxy Server, Redirect Server and Registrar. An end point entity is called as user agent.
User Agents initiate and terminate sessions by exchanging requests and responses.

RFC 2543 defines the User Agent as an application, which contains both a user agent client and user
agent server as follows:
User Agent Client (UAC): a client application that initiates SIP requests.
User Agent Server (UAS): a server application that contacts.

To update a location database with the contact information of the user specified in the request, a server
which accepts REGISTER requests is called as Registrar. Proxy Server is an intermediary entity that acts
as both a server and a client for the purpose of making requests on behalf of other clients. a server that
accepts a SIP request, maps the SIP address of the called party into zero (if there is no known address) or
more new addresses and returns them to the client is called as redirect server. Request-Response type
messages of SIP are shown in figure 1.

98 | 2015, IJAFRC All Rights Reserved www.ijafrc.org

 International Journal of Advance Foundation and Research in Computer (IJAFRC)
Volume 2, Issue 5, May - 2015. ISSN 2348 4853

Figure1: SIP Session Establishment and Call Termination

III. METHODOLOGY
This section discusses a system model for the performance analysis of VoIP codecs. For this purpose,
opnet simulation 14.5 tool is used. It provides a comprehensive development environment supporting
the modeling of communication network and distributed systems.[4] Opnet modeler provides better
environment for simulation, data collection and data analysis. Three scenarios for each codec are created
and set. One SIP proxy server and a no. of SIP clients and servers in each side are created in each
scenario. To run VoIP over SIP clients and server two items are added and configured accordingly. An
application configuration and profile configuration is also set. The encoder scheme is the parameter of
VoIP used here for the simulation process. The signaling protocol is SIP(Session initiation protocol) and
voice frame per packet is set to 1. The SIP UAC Service (User Agent Client Service) requires to be enabled
in each SIP client. It is also necessary for each SIP UAC (caller) to specify the Proxy Server Address to
which SIP requests are sent. Two local networks are included in both sides of backbone network and
each one has a SIP server ,an Ethernet machine supporting VoIP and the signaling protocol is SIP which is
also used by an IP phone. To handle traffic a subnet of 10 nodes are added and all elements in simulation
are connected to the 100 Base-T link. Routers connected by PPP_DS1 are also included in our backbone
network. The application definition support three application i.e VoIP application, FTP, and video
application. Figure 2 shows VoIP application definition.

Figure.2: VoIP application definition

There are three applications included in profile configuration and those are FTP, video and VoIP
application. Figure 3 shows the detail of VoIP profile definition.

99 | 2015, IJAFRC All Rights Reserved www.ijafrc.org

 International Journal of Advance Foundation and Research in Computer (IJAFRC)
Volume 2, Issue 5, May - 2015. ISSN 2348 4853

Figure 3: VoIP Profile definition

IV. RESULTS AND DISCUSSIONS

The result of defined scenario is given in this section. Delay, jitter and traffic (sent and received) are the
QoS parameters for the performance analysis. Delay or latency is defined as the time required for a frame
or a packet to travel from the source to its final destination.[10]The main source of delay is categorized
into: Transmission Delay and destination processing delay, capacity calculation ineffective or insufficient,
technological constraints, reordering packets queuing delay etc. Figure 4 shows the simulation result of
packet delay variation of three codecs:-

Figure 4: Packet Delay variation

From the graph, we may conclude that the voice codec G.729 shows minimum voice packet delay
variation and the voice codec G.711 shows the maximum delay variation than the other two codecs. The

100 | 2015, IJAFRC All Rights Reserved www.ijafrc.org

 International Journal of Advance Foundation and Research in Computer (IJAFRC)
Volume 2, Issue 5, May - 2015. ISSN 2348 4853

delay variation of voice codec G.723.1 lies in between the voice codec G.729 and G.711. Hence, a better
result is given by the voice codec G.729. Figure 5 shows the simulation results of jitter:-

Figure 5: Jitter

The variation of latency is jitter. An absolute value of delay difference between selected packets to arrive
at destination is called as jitter. No jitter means a network with constant latency and no modification. The
variation of jitter for audio codec G.711, G.723 & G.729 is shown in the graph. The amount of data that
can actually be transmitted over the communication channel is called throughput. It is used to estimate
the efficiency of network. Figure 6 shows the traffic received for each voice codec.

Figure 6: Traffic Received

The traffic received for voice codec G.711 is maximum than the other two codecs. The traffic received for
voice codec G.723.1 lies in between the voice codec G.729 & G.711.

V. CONCLUSION

Recent advances in Internet Technology have changed the way people communicate. The voice over
internet protocol(VoIP) is an interesting technology for voice applications over internet. Voice calls can
now be made with a better communication quality and less cost than PSTN. In this paper, the
performance of various VoIP codecs is analyzed in terms of some QoS related parameters using Opnet
Modeler 14.5. In general it is important that voice over IP should deliver maximum throughput, minimum
101 | 2015, IJAFRC All Rights Reserved www.ijafrc.org
 International Journal of Advance Foundation and Research in Computer (IJAFRC)
Volume 2, Issue 5, May - 2015. ISSN 2348 4853

delay and jitter value. VoIP codecs G.711,G.723 and G.729 were simulated so that we may find the most
appropriate one. From our simulation results, we can conclude that the VoIP codec G.723 and G.729
shows minimum jitter value. The voice codec G.729 is best in the terms of delay variation. For future
study, various encoders must be investigated to observe the performance of VoIP and the QoS
parameters should be improved to get the maximum throughput, minimum delay and jitter.

VI. REFERNCES

[1] F. Rahmat, F. Idris, M. Azri, V. Kanathasan, N. Sarimin, and M.H. Mohamad, Performance Analysis
of VoIP in Multi-Hop Wireless Network
[2] GysberthMauritsWattimena, Analysis Performance VoIP Codecs overWiMAX Access Network,
International Journal of Advanced Research in Computer Science and Electronics Engineering
(IJARCSEE)Volume 1, Issue 7, September 2012
[3] HazriRaziff Othman, Darmawaty,M. Ali, NurulAnisMohdYusof, Ku SitiSyahidah Ku Mohd Noh,
AzlinaIdris,Performance Analysis Of VoIP over Mobile WiMAX(IEEE802.16e)Best-Effort Class,
IEEE 5th Control and System Graduate Research Colloquium, Aug. 11 - 12, UiTM, Shah Alam,
Malaysia
[4] Jinia, Jarnail Singh, Ashu Gupta, Analysis of VoIP by varrying the number of nodes failure in
WiMAX Network
[5] Khamis AlAlawi, Hussain Al-Aqrabi, Quality of Service Evaluation of VoIP over Wireless
Networks
[6] Muhammad Aamir, Syed Mustafa Ali Zaidi, QoS Analysis of VoIP Traffic for different Codecs and
Frame Counts per Packet in Multimedia Environment using OPNET
[7] RohaniBakar, Muhammad Ibrahim and D.M. Ali, Performance Measurement of VoIP over WiMAX
4G Network, 2012 IEEE 8th International Colloquium on Signal Processing and its Applications
[8] SheetalJadhav, Haibo Zhang and Zhiyi Huang, Performance Evaluation of Quality of VoIP in
WiMAX and UMTS, 2011 12th International Conference on Parallel and Distributed Computing,
Applications and Technologies
[9] S.Nithya, Dr.M.S.K.Manikandan, Perform ace Analysis Of CODEC's With QoS Constrainsts In Voice
Over Internet Protocol V6
[10] Tarik ANOUARI, Abdelkrim HAQIQ, Comparative Study and Analysis of VolP traffic over WiMAX
using different service classes, Next Generation Networks and Services NGNS, 2-4 December
2012 Portugal
[11] Younes LABYAD, Mohammed MOUGHIT, Abdelkrim HAQIQ, Performance Analysis and
Comparative Study of Voice over IP using Hybrid Codec.

AUTHORS PROFILE

Meenakshi Chawla, Assistant Professor (CSE & IT Dept.) in The Technological Institute Of Textile & Sciences,
Bhiwani. B.Tech(IT) from TIT&S, Bhiwani And M.Tech(IT) from Banasthali University (Banasthali).

102 | 2015, IJAFRC All Rights Reserved www.ijafrc.org

 International Journal of Advance Foundation and Research in Computer (IJAFRC)
Volume 2, Issue 5, May - 2015. ISSN 2348 4853

Priyanka Grover, M.Tech(CSE) student in The Technological Institute Of Textile & Sciences, Bhiwani(Haryana)

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