The key QoS parameters for VoIP are delay, jitter and loss. In the Internet, VoIP requires the underlying packet switched network to minimize the impact of these parameters. A major contributing factor in this regard is traffic engineering carried out by scheduling algorithms. This paper studies the behavior of different types of scheduling algorithms on the delay, jitter and loss QoS parameters. The performance evaluation involves identifying the scheduling algorithms which are most suitable for VoIP communications. The result from the analysis also shows the impact of the QoS parameters on VoIP over the Internet.

1. International journal of Computer Networking and Communication (IJCNAC)Vol. 1, No. 2(November 2013)
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A Comparative Analysis of the Performance of
VoIP Traffic with Different Types of Scheduling
Algorithms
Damilola Osikoya1, KasinathBasu2
Department of Computing ,Oxford Brookes University,Oxford, UK
branco22000@yahoo.com
Department of Computing ,Oxford Brookes University, Oxford, UK
kbasu@brookes.ac.uk
Abstract
The key QoS parameters for VoIP are delay, jitter and loss. In the Internet, VoIP requires
the underlying packet switched network to minimize the impact of these parameters. A
major contributing factor in this regard is traffic engineering carried out by scheduling
algorithms. This paper studies the behavior of different types of scheduling algorithms on
the delay, jitter and loss QoS parameters. The performance evaluation involves
identifying the scheduling algorithms which are most suitable for VoIP communications.
The result from the analysis also shows the impact of the QoS parameters on VoIP over
the Internet.
Keywords:QoS Parameters, VoIP, Scheduling Algorithm, Internet
1
INTRODUCTION
Voice over IP (VoIP) is a telecommunication service that uses packet switched network
infrastructure such as the Internet to facilitate interactive voice communication using telephone or
computer, [8]. This approach is significantly different from traditional telecommunication service
which is based on a circuit switched infrastructure. The later approach is more reliable, but
significantly more expensive and wasteful in terms of resources. VoIP in contrast uses existing
packet switched networks to support VOIP calls, [7]. However, since packet switched network
was initially designed mainly to support traditional data traffic, it lacks any inherent support to
facilitate the real-time and interactive nature of the voice calls. Therefore, traffic engineering of
the existing packet switched network is essential to support VoIP.
To a user, QoS in voice over IP is an attempt to get the best possible quality of voice in a call.
It is a measure of the quality of service delivered to a user, [7]. The performance of a network can
be characterized by a set of parameters called Quality of service (QoS) parameter. Some of the
key QoS parameters for VoIP include delay, jitter, loss and error. It is important that a network
can support these QoS requirement in order to successfully provide a VoIP service.
One of the main challenges of a network that affects the QoS is congestion. Congestion is
caused due to limited resources such as link bandwidth, processor capacity and buffer space. This
results in bottleneck in the forwarding devices such as routers resulting in delay, jitter and loss.
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Congestion can be handled by appropriate admission control policy along with optimizing the
usage of various resources within a network. One significant area of optimization is the
scheduling algorithms used within the routers.
The use of appropriate scheduling algorithm based on the nature of traffic can reduce
congestion and achieve significant improvement in the QoS performance of the network. There
have been several proposed scheduling schemes for supporting the QoS requirements of VoIP
traffic over a packet switched network. Some of the most prominent algorithms include Weighted
Fair Queuing (WFQ), Priority Queuing (PQ) and Custom Queuing (CQ). This paper presents a
comparative analysis of the performance of these scheduling algorithms on the main QoS
parameters of VoIP traffic.
The paper is organized as follows: Section 2 gives a brief insight on QoS mechanism for VoIP
networks; Section 3 provides the literature review; Section 4 presents the brief discussion on
traffic characteristics of VoIP focusing primarily on the key QoS parameters; Section 5 describes
the simulation set-up and VoIP traffic attributes; Section 6 presents an analysis of the simulation
results; and finally Section 7 presents the conclusion of this research.
2
QOS MECHANISM FOR VoIP
Different mechanisms provide QoS for IP networks. Because the Internet connects multiple
domain such as Autonomous systems, the integration between two domain is very important in
achieving proper end-to-end QoS Delivery [14]. Although there are different frameworks for
implementing QoS, the two main architectures are; Integrated Services (IntServ) and
Differentiated Services (DiffServ).
2.1
IntServ
Integrated service (IntServ) is an architecture developed by IETF and is based on per flow
resource reservation. It requires that an application must first make reservations before it is able
to transmit to the network. To make this reservation, the Internet reservation protocol RSVP is
required. IETF RFC2210 describes both the IntServ architecture and the RSVP application. To
reserve bandwidth in a network, RSVP over a system where the user requests QoS for each
session, . In a voice session, the SIP client sends an RSVP path message to the receiver. Each
node along the path on receipt, verifies the message by checking its resources before sending, [3].
As the message reaches the user, a message is sent back to the sender through the network.
2.2
DiffServ
DiffServ is an internetwork architecture described in IETF RF C2474 [10] and updated in RF
C3260 [14]. DiffServ provides QoS, [13], by enforcing policies in a network to provide Service
Level Agreements in the network. It also uses a stateless approach to reduce the use of nodes in a
network [10]. DiffServ is a per-aggregate-class service that uses packet tagging, [3]. It also uses
Type of Service (ToS) flag in the IPv4 header which matches with the flag in the IPv6 ag.
DiffServ classifies network classes and gives each packet a separate treatment based on the
settings of the ToS flag. It provides a Bandwidth Broker (BB) for resource allocation and also
makes sure that network resources are evenly provisioned [2].
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3. International journal of Computer Networking and Communication (IJCNAC)Vol. 1, No. 2(November 2013)
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MPLS
MPLS offers a very good potential for integrating quality of service. It is used to support voice
services in a VoIP network, [3]. MPLS can be used to establish label switching paths between
ingress and egress points, thereby creating a tunnel for labeling traffic. Transmitting traffic over a
label switched path guarantees it`s delivery as long as the allocated bandwidth is not exceeded.
When a packet ingresses into the network through a label switching router (LSR), the router
determines the label switching path (LSP) to use as it adds a label to the path. When a packet is
forwarded, the intending router receives it and decides on the outgoing interface and label value
based on the incoming interface and label value, [3].
3
RELATED WORKS
This paper looks into how different scheduling algorithms behave and their effect on delay,
jitter and loss QoS parameters. The performance evaluation involves identifying the scheduling of
algorithms which are most suitable for VoIP communications. We have used OPNET to perform
the experiments and analyzed the results based on the impact of the QoS parameters on VoIP over
the Internet.
There are several related works that have also being carried out by other researchers in this field
including [6], [9], [11], [4]. While [6] and [11] focused on the analysis of VoIP parameters over
voice traffic for transport protocol support and efficiency, [9] focuses on the factors a affecting
the growth of VoIP.
As discussed by [3] the problems associated with QoS for VoIP networks they propose a
framework that provides a base on which the quality VoIP networks can be built using an MSF
approach. This approach is designed to assist vendors and operators in deploying QoS capable
VoIP networks. Other authors, [2] provided a detailed insight into implementing QoS frameworks
such as IntServ over DiffServ networks for providing QoS on VoIP applications.
This work is a simulation based experiment. The simulation tool used for the experiment is
OPNET network simulator.
4
TRAFFIC CHARACTARISTICS IN VOIP
VoIP exhibits certain properties that affect the quality of voice delivery over a network. These
properties primarily include delay, jitter and loss, [12]. Delay is the time taken for a packet to
travel from source to destination. QoS is assured in VoIP when delay is less than or equal to
150ms. However, it is still acceptable if delay is between 150ms and 400ms. Delay can be
reduced by using appropriate codec and queuing algorithms. It can also be reduced by sending
fragmented packets over the network and reassembling them at the destination. This however
increases the overhead load on the network device.
Jitter in VoIP is a variation in delay. It is the difference between the minimum and the
maximum end-to-end delay and signifies the variable delay within the network. Packets
transmitted at constant rate are expected to be received at constant rate. However, due to network
conditions packets may arrive at variable rate. To balance jitter, a jitter-buffer is used to tailor out
all packets received at irregular intervals and shape them to constant intervals before the packets
are processed.
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Apart from delay and jitter, packet loss and error also affects the quality of voice delivery.
Packet loss is primarily due to buffer over flow within the router`s memory but could also be as a
result of bad transmission, late delivery or general network errors. Unlike in data applications
where lost or errored information can be retransmitted, in VoIP retransmission delay is
unacceptable and therefore traffic engineering policies has to be implemented to limit loss to a
minimal level. In this context, scheduling algorithms have an important role to prioritize and
forward the real time traffic over traditional data traffic.
Other mechanism such as resource reservation, admission control and active queue
management could also reduce the level of packet loss.
5
SIMULATION SETUP
The simulation was setup using the OPNET network simulator. The simulation environment
consists of network components communicating with 2 Ethernet gateways located on two office
floors which act as routers. The first floor of the office consists of two VoIP client workstations,
one FTP client and one video client. The second floor of the office consists of four workstations,
three of which act as VoIP clients
Figure 1: VoIP and Video over an Office Network
and the fourth act as video server and one FTP server.
Both routers are connected to each other via the PPP DS1 cable. The PPP DS1 connects two
systems running IP and it has a data rate of 1.544Mbps. Other devices are connected to the
routers with the 10 Base T cables which transmit data at 10Mbps and have a maximum length of
100 meters. The model includes an application definition, a QoS definition and a problem
definition. The application definition consists of a list of the applications to be run on the network
such as VoIP applications, FTP applications, and video applications.
The applications send traffic into the network which are tested and analyzed. The QoS
definition characterizes protocols that are supported at the IP layer. It defines different scheduling
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5. International journal of Computer Networking and Communication (IJCNAC)Vol. 1, No. 2(November 2013)
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algorithms such as FIFO, WFQ, Custom algorithm and PQ. The pro le configuration is used to
create user profiles. It can be specified on different network interfaces to create traffic on the
application layer. Since this simulation focuses on Quality of Service in VoIP, hence the QoS
profile has been included in the simulation and different scheduling algorithms have been set for
the study
5.1
VoIP Application Attribute
The experiments are based on G.711 and G.729 encoder. The G.711 encoder is based on
uncompressed Pulse Code Modulation (PCM) voice and could be used to encode straight from a
traditional telephone network, [5]. The G.711 use 8 bits per sample and each sample is generated
every 125 microseconds with the use of PCM and this leads to a bit rate of 64 Kbps. The G.729
generates compressed voice and typically operates at 8kbps and is ideal for VoIP because of its
low bandwidth.
Voice speech consists of talk spurt length and silence length which have default values. Talk
spurt is defined in VoIP as the length of sound in-between silence period. The Type of Service
(ToS) field value is set to interactive voice. The G.711 voice frame used in OPNET is 32 bytes
long and therefore of 4 milliseconds duration. The first set of experiments from Section 6.1 to 6.4
is based on using one G.711 frame per packet whereas Section 6.5 is based on 20 frames per
packet. Section 6.6 describe G.729 encoder scheme with 1 and 80 voice frames per packet.
6
SIMULATION RESULTS
The simulations result demonstrates the impact of scheduling algorithms on the main QoS
parameters (delay, jitter and loss) of VoIP. Although these algorithms have their own advantages
and disadvantages, the focus of the experiments has had an impact on the quality and
performance of VoIP traffic. Different types of VoIP traffic have been analyzed with FIFO,
PQ, WFQ and CQ scheduling algorithms. Each of the experiments was run for ten minutes of
simulation time and the following parameters were observed: IP packet loss, point to point delay,
end-to-end delay, and delay variation.
The G.711 encoder scheme was used as a standard for the first set of experiments. At the next
stage another set of experiments were performed using the G. 729 encoder. In each case, we
compared the effect of the scheduling algorithms on the QoS parameters.
Table 1: G.711 Encoder Scheme
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6.1
Packet Loss
Figure 2: Point to point queuing delay and Dropped traffic (packets/second)
Figure 2 shows the performance of the four different scheduling algorithms during a period of
congestion at the router. As shown in the figure above (Fig. 2), both FIFO and CQ suffers sever
packet loss after around2minutes into the simulation as packets begin to get queued up in the
buffer. The drop rate increases with time with CQ showing worse performance than FIFO.
In the case of PQ and WFQ there is no loss till about six minutes and after which there is
occasional loss in rare instances. Therefore, it can be concluded that if packets enter the network
according to their assigned weights and priority as in the case of PQ and WFQ, then congestion
and loss can be significantly minimized.
6.2
Point to Point Queuing Delay
Figure 3 shows the point-to-point queuing delay between floor 1 and floor 2 routers. The two
routers are connected with the PPP DS1 with a data rate of 1.544 Mbps. In all the cases, delay is
not significant
Figure 3: Point to point queuing delay
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7. International journal of Computer Networking and Communication (IJCNAC)Vol. 1, No. 2(November 2013)
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and is constant in the range 0.00250 to 0.00270 between the period 2 and 10 minutes of
simulation time. Therefore, the scheduling algorithms have minimal effect on the point-to-point
delay of the network
6.3
End-to-End Delay
Figure 4: Voice packet end-to end delay
Figure 4 shows the effect of different queuing algorithms on the end-to-end delay. For FIFO,
delay increases exponentially from about 1 minute and 40 seconds into the simulation after which
it remains constant at 1.28 seconds. This means that as the queue builds up, the buffer
experiences delay. Since packets are queued according to their order of arrival and FIFO offers
best e ort service, delay is experienced when incoming VoIP traffic is held back by other types of
traffic. There may also be increase in delay when a particular application hogs the whole buffer
preventing flow of traffic.
The red dotted line shows the graph for PQ. Using the PQ algorithm, packets are scheduled
according to their assigned priority. This reduces the effect of delay in the buffer as VoIP packets
are prioritized over other data packets. The above graph shows that delay is constant for PQ just a
little above zero mark. Custom Queuing behavior was identical to priority queuing with packets
experiencing minimal delay. After the simulation, no result was generated for WFQ which shows
that with WFQ the VoIP packets did not experience any delay
6.4
Delay variation
Figure 5 shows the impact of the various queuing algorithms on the packet to packet delay
variation. In all the cases, initially there is no jitter till 1 minute and 40 seconds into the
experiment as the queuing
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Figure 5: Voice Packet Delay Variation
buffers are empty. For WFQ and CQ jitter stays at zero throughout the experiments which shows
that there is no significant delay variation with both the algorithms.
For PQ, there is a minimal jitter thereafter but its stays constant throughout the remaining part
of the experiment. In the case of FIFO queue the VoIP are treated on a best-e ort basis and
therefore jitter becomes very prominent after the initial period. The result shows an exponential
increase thereafter which however falls and stabilizes after a period.
6.5
Analysis with the G.711 Encoder Scheme
The above sets of experiments were based on using one voice frame per packet. This section
presents the results of rerunning the same experiments with 20 frames per packet.
Figure 6: Delay Variation
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Figure 6 shows the effect of jitter on the four different queuing algorithms with G.711 encoder
scheme and 20 voice frames per packet. In G.711 with1 voice frame per packet the worst case
jitter value for FIFO was 0.16 second; here the same jitter has risen to 1.20 signifying that an
increase in the number of voice frames per packet would also increase the jitter in the network.
This is due to the fact that more packets would have to be processed. Priority queuing also
experiences a small amount of jitter, but custom queuing and WFQ shows no significant delay
variation according to the graph.
In the case of end-to-end delay for G.711 encoding scheme, the delay for the FIFO algorithm
rises exponentially and then stabilizes. This is partly identical to the trend in section 4.3 with 1
frame per packet. The delay for CQ, WFQ and PQ remains negligible.
Figure 7: End to End Delay
6.6
Analysis with the G.729 Encoder Scheme
These set of experiments shows the point-to-point delay between the first floor and second floor
routers using G.729 encoder with 1 and 80 voice frames per packet.
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Figure 8: G.729 with 1 voice frame per packet
Figure 9: G.729 with 80 frames per packet
Figure 8 and 9 shows the point to point delay with 1 and 80 frames per packet. There is less delay
in the G.729 with 1 voice frame per packet compared to 80 frames per packet. In the former case
the delay is 0.004 seconds whereas in the latter case it is 0.0065.Therefore, the point-to-point
delay increases between the first floor router and second floor router when the number of frames
per packet increases.
7 CONCLUSION
VoIP is rapidly replacing the traditional PSTN. However, even with its rapid growth, flaws such
as congestion which results in dropped packet, delay and jitter and a affects the quality of the
voice communication needs to be resolved before VoIP can fully take over the telephony world.
This paper presented an analysis of the queuing algorithms on the main QoS parameters of
VoIP traffic. From the analysis, it is evident that WFQ, PQ and CQ offer better QoS to VoIP.
These scheduling algorithms significantly minimize delay, jitter and loss of the VoIP traffic.
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The paper also demonstrates that the performance of the scheduling algorithms is not affected
by the nature of the VoIP codec viz. G.711 and G.729. Similarly, the number of frames per packet
has minimal effect on the WFQ, PQ and CQ algorithms.
It is however evident from the experiments that FIFO is not suitable for VoIP and is very
sensitive to the overall traffic load in the network during period of congestion. It is therefore
recommended that priority queuing, weighted fair queuing and custom queuing should be used
for handling voice traffic over the Internet.
8
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