This full text paper was peer reviewed at the direction of IEEE Communications Society subject matter experts

for publication in the ICC 2007 proceedings.

MOS-Based Rate Adaption for VoIP Sources

N. T. Moura*, B. A. Vianna*, C. V. N. Albuquerque, V. E. F. Rebello and C. Boeres
Instituto de Computação – Universidade Federal Fluminense – Brazil
{nmoura,bvianna,celio,vinod,boeres}@ic.uff.br

Abstract—This paper proposes an algorithm for the adaptive
adjustment of the transmission rate of VoIP sources based on
the voice quality estimated at the receiver. This adjustment is
achieved through the appropriate use of differing voice codecs,
as the conditions of the network change, in order to maintain
an efficient utilization of the available resources. To validate
our proposal realistically, we have made an effort to simulate
VoIP calls using sources that follow Brady´s model of human
conversations. We investigate the effects of the proposed model
on the aggregate network traffic and compare the results with
existing related work. Simulation results show that the
proposed algorithm makes better use of the available
bandwidth, achieving superior performance in comparison to
similar works.
I. INTRODUCTION
that the Internet is based on a best effort model and does not
reserve resources, it is prone to suffer from congestion and
thus loss of packets).
Transmission of voice over data networks is possible
through different voice coding techniques. These codecs
generate a constant data rate during the speech periods,
independent of the network conditions. Note that the choice
of voice codec used by an application will limit the
maximum attainable QoS level. Moreover, as network
conditions worsen, so will the QoS. Existing applications are
generally hardcoded with a specific codec to provide a
reasonable QoS under typical average network conditions.
Given that these conditions can vary abruptly [19], the QoS
experienced by users is significantly worse than expected.
One approach to solve this problem is to use applications

V OIP (Voice over Internet Protocol) is an increasingly
popular service for voice calls over IP networks. This
includes signaling for establishing and completing each call,
as well as digitalizing, coding and packing the voice signal
so that it can be transmitted by the data network. While VoIP
services are commercially attractive due to their low cost,
their success will be influenced by consumer satisfaction, in
relation to the quality of the calls, and how closely this
quality compares to that of conventional fixed or cellular
telephone services. In addition to economic motivations, the
integration of data and telephony networks makes it possible
to offer new services (e.g. voice mail, instant messaging,
conferencing, etc.). However, the inability of initial VoIP
implementations to cope with the unpredictable nature of the
Internet, seriously affected the acceptance of early VoIP
telephone services.
Given the Internet is based on the best effort service
model, there are no guarantees about the delivery nor the
delay experienced by packets while traveling through the
network. Thus, it is a challenge to have VoIP services offer
the same quality of service (QoS) as a conventional
telephony network, i.e. reliable and with quality of service
guarantees. QoS for VoIP services has been at the center of
attention in numerous research efforts [4], [5], [6], [10], [11],
[12], [15] and [20].
Three main factors affect the perceived voice quality at
the receiver: the total end-to-end delay which depends on the
VoIP architecture, and is influenced by a variety of
parameters such as packet size, coding algorithm, playout
buffer size and network characteristics (e.g. latency,
bandwidth, network topology and configuration); the effect
of delay variability (or jitter); and the packet loss rate (given
* feedback-based rate adaptation for interactive voice
Supported by post graduate scholarship from CAPES.
1-4244-0353-7/07/$25.00 ©2007 IEEE
628
that try to match the transmission rate with the available
network capacity. This adaptation can be made by simply
changing the codec in use.
The objective of this work is to propose a new algorithm
for transmission rate adaptation at VoIP sources based on the
voice quality. The algorithm at each source obtains feedback
information from its corresponding receiver and makes
decisions that aim to maximize the voice quality perceived at
receiver. In this paper, this quality is reflected by a MOS
(Mean Opinion Score) value (described in Section IV)
estimated from statistics of the network and architecture
behavior during a VoIP call.
This paper is organized as follows: Section II presents
some related work while Section III describes the
architecture adopted to model VoIP conversations in a
realistic fashion. Section IV presents the proposed MOS
approach for monitoring the quality of VoIP calls. Section V
outlines a new algorithm for transmission rate adaptation,
adaMOS, which takes into consideration the perceived voice
quality. Section VI reports on some of the simulations
performed. The results obtained demonstrate the good
performance of adaMOS in different scenarios. Section VII
presents final considerations, conclusions and future work.
II. RELATED WORK
Adaptive sources, based on dynamic rate control with
feedback, for multimedia flows have been studied for some
time [2], [17], [23] and [25]. This class of mechanism tries to
match the transmission rate with the available network
capacity, aiming to minimize congestions. Results show that
adaptive applications are more robust and efficient in
presence of congestion, and thus able to transmit more audio
streams while maintaining an acceptable QoS level.
In contrast, relatively little work has been carried on
 This full text paper was peer reviewed at the direction of IEEE Communications Society subject matter experts for publication in the ICC 2007 proceedings.
communication such as VoIP. Barberis et al. [1] proposed an
adaptive transmission rate algorithm based on network
measurements, with the intention of detecting temporary
congestions and estimating the network’s bottleneck link
capacity. Based on this information, the adaptive algorithm
selects a transmission rate compatible with the network
capacity. Simulations show that the algorithm is able to react
rapidly to changes in the network state, but can arrive at the
wrong conclusion. For example, increases in network delay
do not necessarily harm voice quality. It is known that for a
total end-to-end delay in the range from 0 to 150 ms, the
MOS shows little variation [22].
A new, but relatively simplistic, quality of service control
scheme was proposed by Qiao et al. [18] for use in a
differentiated service network (DiffServ) [11] by combining
a adaptive transmission rate strategy (through the use of
Adaptive Multi-Rate codec (AMR) [7]) and the marking of
voice packets with a higher priority. The algorithm,
however, assumes that all sources transmit at the same rate
and, when a rate adjustment is performed, it occurs
simultaneously at all sources. This constitutes a serious
limitation, since it assumes that all communication share the
same network path and an implementation of such a system
requires a global controller, which is not practical in IP
networks. Another contribution of their work was the
adaptation of the transmission rate based on an estimated
MOS observed at the receiver. However, as discussed in
Section IV, it is unclear how closely this estimated MOS
(which only considers packet loss [24]) approaches the real
MOS. Furthermore, their architecture does not use a playout
buffer to reduce the effects of network jitter.
In contrast with existing rate adaptation mechanisms, this
paper introduces a new algorithm for transmission rate
adaptation for today’s Internet and validates this proposal in
a simulated environment. The adaptation is completely
distributed and independent for each VoIP communication.
In addition to considering packet loss, this paper highlights
the importance of considering the network delay, end-to-end
delay and jitter, to obtain a more accurate voice quality
measure.
III. THE ARCHITECTURE
Fig. 1 illustrates the architecture proposed for the adaptive
control of the VoIP source coding rate introduced in this
work. At the source, the voice is digitalized, encoded and
packed to be transmitted by the network. RTP/UDP/IP
protocols are then responsible for delivering the voice
packets to their destination. When arriving at the receiver,
the jitter is removed by a playout buffer and packets are then
decoded. At the receiver’s side, information about delay, loss
and estimated MOS is collected and returned to the source,
where the proposed adaptive transmission rate control
algorithm (adaMOS) will determine the appropriate
transmission encoder to maximize the voice quality
perceived by the receiver. In the following sub-sections the
main elements of this architecture will be described in detail,
namely: the VoIP source, the VoIP receiver and the feedback
mechanism.
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IP
Network
Source
Coder
Rate Controler (adaMOS)
Via RTCP Decoder
IP
Network
Feedback Information
Receiver
Fig. 1. The proposed adaMOS architecture
A. The VoIP Source
Another limitation observed in the works cited in Section
II is the representation of voice flows as continuous data.
Given that human conversation consists of alternating speech
and silence periods, it is more appropriate for sources to
employ silence suppression techniques and only transmit if a
speech period is detected. This on-off behavior can have a
significant effect on aggregate voice traffic. In [3], Brady
showed that the speech and silence periods can be
approximated by an exponential distribution with an average
of 1/λ (ON) and 1/µ (OFF). λ-1 = 1.004s and µ-1 = 1.587s, are
typically assumed values, which include the hangover time
(i.e. any period of silence with a duration less then 200ms is
considered part of a speech period). In this paper, the use of
sources that reflect this voice model is considered and the
effect of aggregated network traffic from multiple sources is
investigated. Each source employs the proposed adaMOS
adaptive algorithm to adjust its transmission rate.
B. The VoIP Receiver
Each of receivers used in the proposed architecture
employ a playout buffer with the objective of minimizing the
effect of jitter on the voice quality. Two adaptive playout
buffer techniques were studied and implemented in the
simulation environment in order to model receivers more
realistically in the proposed architecture and to evaluate the
behavior of playout buffers in face of VoIP sources that
adjust their transmission rates adaptively. The first technique
is based on the work in [19], where the playout delay (the
time between when the packet is generated and played) is
adjusted only between talk spurts. This mechanism is
referred to hereinafter as a talk spurt buffer. The second
variant is a technique where the playout delay is adaptively
adjusted, in accordance with the network conditions, for each
individual packet even during a talk spurt as proposed in
[13]. This mechanism is referred to as a packet playout
buffer in the remainder of this text.
C. The Feedback Mechanism
The receivers periodically send feedback information (in
the form of a feedback packet) to their respective sources,
through the RTCP protocol. This information does not
include explicit congestion information related to the state of
intermediate nodes. At the end of each feedback period, each
receiver calculates the arithmetic mean delay experienced by
the packets, the packet loss rate and an estimate of the MOS
 This full text paper was peer reviewed at the direction of IEEE Communications Society subject matter experts for publication in the ICC 2007 proceedings.

experienced within that period. The feedback packet also
includes information on the coding rate of the packets
received during the period. If the feedback period is very
short, there will be a large data flow from the receivers to the
sources. On the other hand, a very long interval will cause
the algorithm to be less receptive to rapid changes in
network conditions and thus compromise the perceived
quality of service significantly. In the future, piggybacking
feedback information in voice packets for bi-directional
VoIP conversations will be investigated.
IV. QUALITY MONITOR
Recent research, such as those cited in Section II, has
focused on VoIP applications that adapt their configuration
to the state of the network with the goal of maximizing the
quality of calls. However, a key issue has been how to
measure voice quality, and therefore design adaptive
applications based on this metric. Earlier VoIP applications
equated quality with information from measurements from
the transport or network layer, e.g. packet loss or network
delay. However utilizing either one of these separately does
not reflect faithfully the quality of service perceived at the
receiver.
Clearly, a good quality of service measure for VoIP calls
must somehow consider the user’s opinion about the service.
The Mean Opinion Score (MOS) is a method recommended
by the ITU-T P.800 to measure speech quality. In this
method, the users rate the call quality in a range varying
from 1 (bad) to 5 (excellent). However, this is a very
subjective method that consumes a significant amount of
time and consequently is both very costly and inappropriate
for on-line adaptations.
Other models aim to determine this score in an objective
way. Amongst them, two are well established: the E-Model
[9] and the Perceptual Assessment of Speech Quality
(PESQ) [21]. It would be interesting to at least be able to
estimate the quality perceived by the users of the VoIP
application, based on quantitative data available, such as the
packet loss rate and/or the delay experienced by packets.
This work employs Equation (1) (shown previously to be a
good approximation for the MOS value [8]) to map network
parameters to an estimated MOS value:
MOS ≈ T − αp + β d − ηd 2 + ϕd 3
end delay and the packet loss rate in the form of an estimate
of the voice quality at the receiver, in order to make
informed decisions regarding changes in the coding rate.
These values are returned to the source through the feedback
mechanism. The basic idea consists of reducing the source
transmission rate when high values of delay, packet loss or a
reduction in the estimated MOS are perceived. Moreover, the
algorithm may increase the transmission rate and thus voice
quality when perceiving that the current network conditions
will support it. Note that this work also employs a playout
buffer to reduce the effects of network jitter and improve
voice quality. To overcome the effects caused by the buffer
delay, the algorithm calculates two different MOS values
using Equation (1): net_MOS, an estimated MOS value prior
to the playout buffer based on the packet loss and network
delay, and; estimated_MOS, an estimated MOS value that
considers the buffer delay and thus reflects the voice quality
experienced by the listener. While net_MOS is used to detect
congestion, or the possibility of it, and thus initiate a rapid
decrease in the codification rate, estimated_MOS is used to
trigger conservative rate increases. It is the combined use of
these metrics that provides adaMOS with a solid foundation
for making decisions. Table I gives a description of the
variables of the adaMOS algorithm presented in Fig. 2.
TABLE I
ADAMOS PARAMETERS AND VARIABLES
Parameter Description
ALFA_ADAPTIVE weighting factor for the smoothed average of the
network delay and packet loss
MAX_LOSS maximum acceptable loss – triggers a rate reduction
MIN_LOSS value at which the algorithm might wish to consider a
rate increase
DELAY_THRESH relative delay value that triggers a rate decrease
(increase) when delay increases (decreases)
INCREMENT_THRESH score required to allow a rate increase – this value
determines how conservative the algorithm is to
transmission rate increases
net_delay average network delay during last feedback interval
loss_avg smoothed loss ratio
delay_avg average (smoothed) delay
decrement flag to indicate a moderate rate decrease
halve flag to indicate a significant (by half) rate reduction
increment indication of the degree to which the network can
support a rate increase
limit_rate the rate that lead to the last rate decrease
backoff_time waiting time before a rate increase
backoff_limit limits the range of values for the backoff_time
adaMOS uses the additive increment/multiplicative
(1) decrement paradigm, similar to some congestion control

where α=0.195, β=2.64x10-3, η=1.86x10-5, ϕ=1.22x10-8, p algorithms such as TCP. It is important to notice that the
is the packet loss rate, d represents the total end-to-end delay multiplicative decrease factor is fundamental for concept of
and T represents the maximum MOS value for a given codec fairness, since systems should aim to distribute bandwidth
(i.e. when the transmission experiences no losses or delays). appropriately among applications. The additive increment is
The objective of the adaptive transmission rate algorithm, a conservative approach for increasing the transmission rate
adaMOS, is to maintain the estimated MOS value at the without causing a dramatic impact on the network.
highest level for the duration of the communication. The This work also considers the use of an exponential backoff
voice quality must be at least acceptable (above 3.6) and, for technique, with intention to prevent the symmetrical
tolling purposes, it is desirable that the MOS remains above oscillatory behavior of the sources. The idea is that a source
4.0 [14]. that suffers successive rate decreases from the same
transmission rate should be prevented from returning to this
V. ADAMOS: THE ADAPTIVE ALGORITHM rate for a certain period of time (the backoff time).
Care must be taken when using the feedback information
The adaptive transmission rate control algorithm, received since it may not yet reflect the effect of a rate
adaMOS, takes into consideration the network and end-to-

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 This full text paper was peer reviewed at the direction of IEEE Communications Society subject matter experts for publication in the ICC 2007 proceedings.

change. Feedback packets are identified as valid (line 2 of minimum delay observed throughout the voice call (line 19)
the algorithm in Fig. 2) if they refer to the current and the estimated MOS experienced an increase (line 20),
codification rate and a sufficient number of the packets this will be interpreted as an improvement in the network
(greater than 30% of the maximum number expected) is conditions, and thus increase the chance of raising the
received during the current feedback interval. One exception transmission rate (line 21).
is when the codification rate during the feedback period is Each time a rate reduction is implemented, the algorithm
lower than the current rate and the packet loss rate exceeds verifies the transmission rate prior to the last rate decrease.
the maximum acceptable loss threshold. This situation If the potential exists for oscillatory behavior (line 29), a
occurs when a recent rate increase probably overloaded the backoff time is chosen to prevent a possible rate increase
network capacity, justifying an immediate rate decrease. from occurring in the near future. This backoff time is
1. //Validating feedback packet determined randomly from range which grows exponentially
2. if ( valid_feedback ) {
3. loss_avg=ALFA_ADAPTIVE*loss_avg+(1–ALFA_ADAPTIVE)*loss_rate;
with the number of successive identical rate reductions (lines
4.
5.
// Check loss rate
if ( loss_avg > MAX_LOSS ) {
30-31).
6. halve = TRUE; The rate increase procedure (described in Fig. 3) is only
7. increment = 0;
8. } called during periods of silence due to the fact that, based on
9.
10.
else if ( loss_avg < MIN_LOSS )
if ( estimated_MOS > previous_estimated_MOS ) simulation results, transmission rate modifications during
11.
12.
increment++;
// Check delay
speech periods harm playout buffer performance, in
13. if ( net_delay > delay_avg * DELAY_THRESH ) { particular the talk spurt buffer. Since estimated playout delay
14. if ( net_MOS < prev_net_MOS ) {
15. decrement = TRUE; (based on the first packet of the spurt) of a talk spurt buffer
16.
17. }
increment--;
is only adjusted between speech periods, any change in the
18.
19.
}
else if ( net_delay < min_delay * DELAY_THRESH ) {
coding rate during a talk spurt may cause the playout buffer
20. if ( estimated_MOS > previous_estimated_MOS ) to discard packets, until the next talk spurt. A rate increase
21. increment++;
22. } only takes effect if a sufficient number of positive
23. delay_avg=ALFA_ADAPTIVE*delay_avg+(1-ALFA_ADAPTIVE)*net_delay;
24. previous_estimated_MOS = current_MOS; indications has occurred (i.e. reached the increment
25.
26.
prev_net_MOS = net_MOS;
// Check decrease rate indication threshold) (line 1) and the previous backoff time has expired
27.
28.
if( halve || decrement ) {
increment = 0;
(line 2).
29. if( limit_rate == current_rate ) {
30. backoff_limit *= 2;
31. backoff_time = (rand() % backoff_limit) + 1; VI. SIMULATION RESULTS
32. }
33. else { This work opts to use the same simple network topology
34. limit_rate = current_rate;
35. backoff_limit = 1; (Fig. 4) used in [1] for the simulated evaluations, since
36.
37. }
backoff_time = 0;
complex topologies, though potentially more realistic, have
38.
39.
if( halve ) {
HalveRate();
the disadvantage of generating results that lead to
40.
41.
}
else if( decrement ) {
inconclusive interpretations. Moreover, the use of this
42. DecrementRate(); topology provides a basis for comparison with the previously
43. }
44. loss_avg = 0.0; cited work. In this topology, the link between switches SW1
45.
46. }
}
and SW2 is the bottleneck link. This link has a configurable
Fig. 2. adaMOS: adaptive algorithm bandwidth (L), and a configurable latency (D). Access links
are assumed to have enough bandwidth for all the traffic
1. if(increment == INCREMENT_THRESH){
2. if((backoff_time>0)&&(current_rate==limit_rate-8000.0)){ traveling from a source (Fi) to a destination (Ri) and present
3. backoff_time=(--backoff_time >= 0) ? backoff_time : 0;
4. } fixed delay of 1ms.
5. else{
6. IncreaseRate();
7. increment = 0;
8. loss_avg = 0.0;
9. }
10. }
Fig. 3. Rate increase procedure

For each valid feedback packet received, the adaptive

algorithm evaluates the network conditions and the estimated
voice quality perceived by the user. If the average loss rate is
greater than the maximum acceptable loss threshold (line 5) Fig. 4. Simulated topology
the current rate is halved (line 6). As well as fairness, the fact
In the following sub-sections, a number of simulation
that packet loss significantly impact voice quality justifies an
scenarios is presented highlighting the performance of the
aggressive reduction. If the average loss rate is below a
algorithm proposed in this work. The NS-2 simulator [16]
given loss threshold (line 9), and the estimated voice quality
has been extended to attend the necessities of our
at the receiver has improved (line 10), an indicator
architecture. All scenarios were simulated for 900 seconds,
representing the network’s ability to support a rate increase
with a feedback interval of 1 second, using Brady´s voice
is upgraded (line 11). If the delay is more than a certain
model and with both adaptive playout buffer techniques. The
threshold above the average (line 13) and the voice quality
results for both playout buffers showed a similar behavior
also suffered a reduction, a rate decrease is signaled (line 14
relative to the estimated MOS, however due to space
to 16). If the delay is within a given tolerance factor of the

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 This full text paper was peer reviewed at the direction of IEEE Communications Society subject matter experts for publication in the ICC 2007 proceedings.

constraints, only the results using a packet playout buffer are is plenty of available bandwidth, high rate codecs are worse
presented. when the available bandwidth is insufficient.
A. Scenario 1: Aggregated Sources 5
adaMOS
AVoIP
In this first scenario, the behavior of 20 simultaneous
VoIP transmissions is evaluated. The results are compared 4
with the algorithm proposed in [1], denoted by AVoIP, for
an identical architecture which included packet playout
3
buffers. Fig. 5 represent the average network delay over the

MOS
20 sources using simulation parameters of L=256kbps and
D=3ms. Notice that AVoIP introduces an oscillating 2
behavior in the network. A possible explanation for this
instability is the use of on-off sources that model the
1
conversation in a realistic way in our architecture. AVoIP
was designed to handle constant bit rate voice flows.
However, the realistic on-off behavior of several aggregate 0
0 200 400 600 800 1000
sources has a negative effect on AVoIP since inappropriate Time (s)
decisions to increase the transmission rate are taken at
Fig. 6. Average MOS quality with 20 sources
specific moments when a number of sources are in the off
state. The adaMOS approach for rate increases is 5
conservative and avoids making rash decisions.
Consequently, as observed in Fig. 6, the perceived voice
quality is maintained stable at a superior and acceptable 4.5
level. In this scenario, adaMOS experienced no packet loss,
while AVoIP incurred an average packet loss of 0.94% and
at certain moments spikes, peaking as high as 58%, were
MOS

4
observed.
adaMOS
120 8kb
adaMOS
AVoIP 16kb
24kb
3.5 32kb
100 40kb
48kb
56kb
64kb
80 AVoIP
3
Network Delay (ms)

0 1000 2000 3000 4000 5000 6000

Bandwidth (Kbps)
60

Fig. 7. Average MOS quality for 100 sources

40
C. Scenario 3: adaMOS Operation
20
This scenario illustrates the voice quality (MOS) of
adaMOS as a function of both the network latency and the
available bandwidth. As expected for 100 simultaneous VoIP
0
0 50 100 150 200 250 300 350 400 450 500 550 600 650 700 750 800 850 900 950 1000 conversations, Fig. 8 shows that high bandwidth low latency
Time (s)
networks provide higher quality connections than low
Fig. 5. Average network delay with 20 sources bandwidth high latency networks. The shaded 2D region
below the plotted surface identifies the range of
B. Scenario 2: Bandwidth Influence latency/bandwidth values for which the voice quality is
This scenario increases the number of simultaneous VoIP acceptable (i.e. MOS of 3.6 or above). For network latencies
transmissions to 100 with the objective of analyzing the under 120ms, adaMOS attains MOS values superior to 3.6,
effects of available bandwidth on the average packet loss and even with a restricted bandwidth of 400kbps for 100 sources.
estimated MOS. The simulation used a main link delay of Fig. 9 compares adaMOS and AVoIP performance for
100ms. Fig. 7 show the playout MOS for 8 codecs without differing bandwidths and network latencies. The area above
adaptation (fixed rate), adaMOS and AVoIP. each line represents again the range of bandwidth and
Most interestingly, notice that when bandwidth is scarce latency combination for which the estimated MOS is
(below 3000kbps) adaMOS causes lower network delays and acceptable. Note that adaMOS achieves the acceptable MOS
loss rates than AVoIP and thus achieves a better utilization value with lower bandwidths and higher latencies when
of the available bandwidth. Also notice in Fig. 7 that the compared to AVoIP. As the available bandwidth increases,
adaMOS quality (estimated MOS) is always above the both algorithms deliver acceptable qualities (higher than
minimum level for acceptability (MOS=3.6 is also a value 3.6).
from where tolling becomes possible). As expected, while
low fixed rate codecs exhibit worse performance when there

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Computer Communications Journal, vol. 24, no. 9, pp. 757–767. May

MOS 2001.
5 [2] Bolot, J., and Vega-Garcia, A. “Control mechanisms for packet audio
4.5
4 in the Internet”. IEEE INFOCOM, pp. 232-239, Apr 1996.
5
4.5 3.5 [3] Brady, P. T. “A model for generating on-off speech patterns in two
3
4
2.5
way conversation”. The Bell System Technical Journal, vol. 48, no.7,
3.5
3 2 pp. 2445-2472, Sep 1969.
2.5
2
1.5 [4] Chen, X., Wang, C., Xuan, D., Li, Z., Min, Y., and Zhao, W. “Survey
1
1.5 0.5 on QoS management of VoIP”. ICCNMC, pp. 69–77, Oct 2003.
1
0.5
0 [5] Cole, R. G., and Rosenbluth J. H. “Voice over IP performance
0 monitoring”. ACM Computer Communication Review, vol. 31, no. 2,
pp. 9–24, Apr 2001.
6000
5000 [6] Ding L., and Goubran, R. A. “Speech quality prediction in VoIP using
4000 the extended e-model”. IEEE GLOBECOM, pp. 3974–3978, Dec
0 50 100 150 3000
Bandwidth (Kbps) 2003.
200 250
300 350
2000
[7] Ekudden, E., Hagen, R., Johansson, I., and Svedberg, J. “The adaptive
1000
Latency (ms) 400 450
500 0
multi-rate speech coder”. IEEE Workshop on Speech Coding for
Telecommunications, pp. 117–119, 1999.
Fig. 8. MOS x Latency x Bandwidth [8] Fujimoto, K., Ata, S., and Murata, M. “Adaptive playout buffer
algorithm for enhancing perceived quality of streaming applications”.
IEEE GLOBECOM, no. 1, pp. 2463-2469, Nov 2002.
3000
adaMOS
[9] ITU-T Rec. G.107, “The E-Model, a computational model for use in
AVoIP transmission planning”, Mar 2003.
2500 [10] Kampichler, W., and Goeschka, K. M. “End-to-end network
performance measurement for voice transmission feasibility”. IEEE
CATA, pp. 501–504, 2001.
2000
[11] Klepec, B., and Kos, A. “Performance of VoIP applications in a
Bandwidth (Kbps)

simple differentiated services network architecture”. EUROCON, pp.

1500
1000
500
0
0 50 100 150
Latency (ms)
Fig. 9. Delay x Bandwidth – adaMOS and AVoIP for a MOS of 3.6
VII. CONCLUSIONS AND FUTURE WORK
This work proposed a new adaptive algorithm for the
transmission rate adjustment of sources for VoIP
applications on the Internet. The principal contribution of the
proposed algorithm is the fact that it not only takes into
consideration network parameters, but also the voice quality
perceived by users unlike existing mechanisms such as
AVoIP [1].
Different from the architecture proposed in [18] (which is
limited to DiffServ networks), this proposed architecture
supports the use of playout buffers. The feedback-based
adaMOS algorithm has been shown to be robust in a variety
of test scenarios. These simulations clearly showed that the
received voice quality is a fundamental criterion for decision
making. Our conservative approach when increasing the
transmission rate is important in many scenarios, especially
if realistic on-off aggregated sources are considered.
The adaMOS algorithm for adaptive rate control of VoIP
sources revealed promising results and future work to
investigate characteristics such as sensitivity, stability and
fairness in scenarios with realistic topologies and interfering
traffic is underway. This will be an important step to certify
the algorithm’s aptitude for the current Internet.
214–217, ACM Press, 2001.
[12] Kos, A., Klepec, B., and Tomazic, S. “Techniques for performance
improvement of VoIP applications”. 11º IEEE MELECON, pp. 250–
254, 2002.
[13] Liang, Y.J., Farber, N., and Girod, B. “Adaptive playout scheduling
and loss concealment for voice communication over IP networks”.
IEEE Transactions on Multimedia, vol. 5, no. 4, pp. 532-543, Dec
200 250 300
2003.
[14] Markopoulou, A.P., Tobagi, F.A., and Karam, M.J. “Assessment of
VoIP quality over Internet backbones.” IEEE INFOCOM, vol.1,
pp.150- 159, Jun 2002.
[15] Muppala, J.K., Bancherdvanich, T., and Tyagi A. “VoIP performance
on differentiated services enabled network”. IEEE ICON, pp. 419–
423, 2000.
[16] NS-2 network simulator 2.29. http://www.isi.edu/nsnam/ns/.
Last visited in Sep. 2006
[17] Perkins, C., Hodson, O., and Hardman, V. “A survey of packet loss
recovery techniques for streaming audio”. IEEE Network, pp. 40–48,
Sep/Oct 1998.
[18] Qiao, Z., Sun, L., Heilemann N., and Ifeachor E. “A new method for
VoIP quality of service control use combined adaptive sender rate and
priority marking”. IEEE ICC, pp. 1473–1477, Jun 2004.
[19] Ramjee, R., Kurose, J., Towsley, D., and Schulzrinne, H. “Adaptive
playout mechanisms for packetized audio applications in wide-area
networks”. 13º IEEE INFOCOM, vol.2, pp. 680 – 688, Jun 1994.
[20] Reynolds, R.J.B., and Rix, A.W. “Quality VoIP - an engineering
challenge”. BT Technology Journal, vol. 19, no. 2, pp 23-32, Apr
2001.
[21] Rix, A., Beerends, J., Hollier, M., and Hekstra, A. “Perceptual
evaluation of speech quality (PESQ)-a new method for speech quality
assessment of telephone networks and codecs”. IEEE ICASSP, vol. 2,
pp. 749–752, 2001.
[22] Savolaine, C. “QoS/VoIP overview”. IEEE CQR International.
Workshop, 2001.
[23] Shenker, S. “Fundamental design issues for the future Internet”.
IEEE/ACM JSAC, vol. 13, no. 7, pp. 1176–1188, Sep 1995.
[24] Sun, L., and Ifeachor, E. “Prediction of perceived conversational
speech quality and effects of playout buffer algorithms”. IEEE ICC,
pp. 1–6, 2003.
[25] Yin, N., and Hluchyj, M. “A dynamic rate control mechanism for
source coded traffic in a fast packet network”. IEEE JSAC, vol. 9, no.
7, pp. 1003–1012, Sep 1991.

REFERENCES
[1] Barberis, A., Casetti, C., Martin, J. D., and Meo, M. “A simulation
study of adaptive voice communications on IP networks”. IEEE

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