IMPROVEMENT OF VOIP QUALITY BY PACKET DROPPING

IN ADSL ROUTERS

Qin Dai, Matthias Baumann

Technische Universität Dresden, Dresden, Germany

Ralf Lehnert

Technische Universität Dresden, Dresden, Germany

Keywords: VoIP, QoS, ADSL Downlink, Queueing Delay, End-to-End Delay, Loss Bursts.

Abstract: Packet dropping is known as a simple mechanism to control TCP traffic. In this paper, TCP packet dropping

is introduced in the egress router of an ADSL downlink. The aim is to improve the quality of VoIP

connections that compete with TCP applications in downlink direction. The ADSL downlink buffer is

assumed to operate as simple FCFS queue. Different simulations have been conducted that evaluate the

mechanism in two scenarios. Firstly, the long-term impact of the mechanism both on VoIP application and

TCP applications is investigated. Secondly, with more realistic network settings, the effectiveness of the

mechanism for a short-time real speech is evaluated. The speech’s PESQ estimate is used to assess the

service quality. The results indicate that in both cases packet dropping can improve the VoIP quality.

However, the required high dropping ratio can result in TCP traffic bursts and therefore unstable VoIP

quality as well as bad TCP performance.

1 INTRODUCTION

As one kind of real-time applications, VoIP (voice

over IP) has its QoS (quality of service)

requirements on delay, delay variation and packet

loss. Among them, delay related measurements are

most critical. The first big problem caused by delay

is the talker overlap. In a two-way conversation it is

hard to continue when both sides start talking at the

same time. The presence of voice echo also has a

significant impact on delay sensitivity. Moreover

those packets which have been delayed for “too”

long time will be discarded by the de-jittering buffer

in the receiver.

VoIP packet delay, which can be caused by

different reasons, can be characterized into two basic

types: the fixed delay and the variable delay. Fixed

delay includes coding / decoding delay, processing

delay, propagation delay, serialization delay etc. All

of these delays are fixed according to the applied

technologies, e.g. the algorithms for coding /

decoding, the bit rate for the packet serialization,

and the propagation delay itself. Variable delay

mainly is caused by varying queue occupations in

network nodes which largely depend on the situation

of competing traffic streams. The queueing delay of

a VoIP packet especially can become serious when

entering a bottleneck link with fast packet stream

aggregation at the input side and slow release at the

output side.

In recent years, ADSL (Asymmetric Digital

Subscriber Line) technology has become one of the

most popular access solutions for broadband

networks. In ADSL access network, voice and data

traffic will be multiplexed / demultiplexed in its

edge node to the IP/ATM network named DSLAM

(Digital Subscriber Line Access Multiplexer). From

the comparatively low speed in the uplink (typically

128Kbit/s), one can expect that QoS problems

mainly arise in this direction, and most of the

research focuses on the uplink problem, e.g.

(Orozco_barbosa, Siddiqui, Yongacoglu, 2002).

Although the downlink speed of ADSL may be quite

large (up to 8 Mbit/s), the considerable bandwidth

difference between core and access network leads to

queueing of downstream packets in the DSLAM. If

the IP network operator does not support different

service types for VoIP and data applications, FCFS

269

Dai Q., Baumann M. and Lehnert R. (2007).

IMPROVEMENT OF VOIP QUALITY BY PACKET DROPPING IN ADSL ROUTERS.

In Proceedings of the Second International Conference on Signal Processing and Multimedia Applications, pages 265-272

DOI: 10.5220/0002135502650272

 SciTePress

(First Come First Serve) queueing may be necessary.

In this paper, we consider such a scenario: different

traffic streams including VoIP, TCP (Web and FTP

traffic) go through the core IP network and pass the

access node DSLAM. The link between DSLAM

and ADSL modem becomes the bottleneck in the

downlink of the communication. The FCFS

queueing in DSLAM leads to unacceptable delay for

VoIP packets.

ITU-T recommendation G.114 states that the

one-way VoIP communication delay should not

exceed 150 ms. We assume a voice communication

with ADSL access on both end points and consider

the downlink performance in one of the points. We

neglect the delay in the Internet and assume that a

non-adaptive de-jittering buffer with the size of 150

ms is used in the receiver side. For ADSL links with

1 MBit/s downlink and 128 kBit/s uplink bit rate, up

to 100 ms of the total delay budget should be

reserved for the remote uplink part of the VoIP

connection. In order to restrict the downlink delay to

the remaining budget, the number of packets queued

by TCP connections has to be controlled.

As one kind of TCP traffic control mechanisms,

packet dropping can effectively limit the TCP

sending rate and has been implemented with

different intentions. To avoid network congestion

and improve the overall performance, Random Early

Detection (RED) uses a linearly increasing function

of the queue length to probabilistically drop

incoming packets with the expectation that the

remote sender can slow down its transmission rate

(Floyd, Jacobson, 1993). (Shyu, Chen, Luo, 2002)

introduces an adaptive dropping mechanism in the

router for Internet congestion control. In this paper,

we implement packet dropping in the ADSL router

at the user’s side and expect that the number of TCP

packets delaying VoIP packets in the DSLAM queue

can be controlled. We investigate the influence of

the mechanism both on the behaviour of the TCP

control loop and on the VoIP application. In our

simulations, we firstly investigate both VoIP and

TCP quality on packet level by using a long time

VoIP session. In the second simulation, we use a

short-time real speech as the VoIP streaming since

the real conversation consists of short speech

sections and silence periods. Additionally, higher

bandwidths both in up and down links are reserved

for the ADSL user in this simulation. A perceptual

evaluation of the VoIP quality will be conducted by

using the PESQ model.

The remainder of this paper is organized as

follows. In Section 2, the packet dropping

mechanism is analyzed and discussed. Later the

PESQ model is introduced. In Section 3, the

simulation models and the metrics for performance

evaluation in two simulations are introduced. The

numerical results are presented in Section 4. Finally

the conclusions are drawn.

2 BACKGROUND

2.1 TCP Packet Dropping

As already mentioned in the last section, it is

necessary to limit the number of packets a TCP

connection can place in the DSLAM buffer. TCP is

a sliding window-based protocol. The TCP sender

restricts its window size according to the minimum

of the receiver’s buffer size and the congestion

window size. In TCP, packet loss is considered as a

sign of network congestion. Reacting on packet loss,

the sender reduces its congestion window to half of

the old value during “Fast Retransmission and

Recovery” phase to reduce network congestion. In

case of retransmission time-out, the window is even

closed to one packet. Thus it seems feasible to avoid

large window sizes by introducing “artificial” packet

losses in the downstream of active TCP connections.

If packets of a single connection are dropped

periodically, the TCP window size cannot exceed a

certain threshold, thus effectively limiting the

maximum number of packets that are placed in the

DSLAM buffer. One drawback of this approach is

that the dropping ratio should depend on the network

parameters, e.g. the number of TCP connections and

the lowest round-trip time (RTT), which can lead to

very low link utilization and bad perceived

performance e.g. for Web applications. Our

simulation results will further illustrate the problem.

Besides it might appear counterintuitive to drop

packets already transmitted successfully over the

ADSL link. In spite of the above disadvantages, the

approach‘s simplicity could lead to an almost gratis

implementation still offering a sensible QoS

advantage.

2.2 PESQ Model

As one objective method for speech quality

assessment, the approach Perceptual Evaluation of

Speech Quality (PESQ) is standardised by ITU-T as

recommendation P.862. Figure 1 shows the basic

philosophy used in PESQ (ITU-T, 2001). Firstly, a

series of delays between original input and degraded

output are computed by the time alignment module.

Secondly, based on this set of delays PESQ

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270

compares the original signal with the aligned

degraded one by using a perceptual model. The key

to this process is transformation of both the original

and degraded signals to an internal representation

that is analogous to the psychophysical

representation of audio signals in the human

auditory system. The differences in internal

representation between the original signal and the

degraded one are computed in the cognitive model.

Finally, an objective listening quality MOS (mean

opinion score) is given. PESQ is believed to be able

to predict subjective quality with good correlation in

a very wide range of conditions, that may include

coding distortions, errors, noise, filtering, delay and

variable delay (Rix, Hollier, Hekstra, Beerends,

2002). The average correlation coefficient between

PESQ and the subjective scores is indicated as 0.935

in (ITU-T, 2001) from 22 known ITU benchmark

experiments. (ITU-T, 2001)

Table 1 shows the ACR (Absolute Category

Rating) opinion scale used in the development of

PESQ (Beerends, Hekstra, Rix, Hollier, 2002). The

speech quality is graded from bad to excellent with

the PESQ score increases from 1 to 5. Since a

comparable voice quality is always expected when

the user switches from the traditional telephone to

VoIP, a speech with low PESQ value can not be

acceptable. Different codecs can introduce different

level of degrading to the voice. (Markopoulou,

Tobagi, Karam, 2003) indicates that the ideal MOS

value of a speech coded with G.729 is around 4. The

PESQ value of 3.5 is taken as the criterion for

judging the quality of speech in our second

simulation, with which the quality of the speech is

considered between fair to good as shown in the

table.

Table 1: ACR listening quality opinion scale used in the

development of PESQ.

Quality of the speech Score

Excellent 5

Good 4

Fair 3

Poor 2

Bad 1

Figure 1: Overview of the basic philosophy used in PESQ

(ITU-T, 2001).

VoIP phone

VoIP

downlink

delay

TCP applications

(web and FTP

clients

)

Ethernet

Web servers

TCP Internet

access node

FTP server

ATM

VoIP

DSLAM

ADSL downlink ADSL uplink

DSLAM buffer

(FCFS)

ADSL router

VoIP

ATM

Figure 2: System model (ATM links have a bit rate of 150 Mbit/s).

IMPROVEMENT OF VOIP QUALITY BY PACKET DROPPING IN ADSL ROUTERS

271

3 SIMULATIONS AND

PERFORMANCE METRICS

3.1 Simulation I

Figure 2 shows the system model adopted in this

simulation. Firstly, we investigate the VoIP quality

using only Web traffic as background load. In a Web

application, usually several short-lived TCP

connections run simultaneously. This leads to bursty

traffic, since the slow start at the beginning of a TCP

connection results in steep packet rate increase. In

our simulation, we use two concurrent Web sessions

to intensify the traffic burstiness. Within each

session, documents from a server farm of 100 Web

servers are downloaded in a random fashion. These

servers are connected to the Internet part of the

network model with link delays ranging from 15 to

35 ms. The traffic model for the Web applications is

taken from (Mah, 1997). It specifies distributions for

the distance between Web requests (exponential), for

the number of objects per main page (Pareto), for the

distance between object requests in a page

(exponential), and for the object sizes (Pareto). In

order to assess the influence of the approach, we

measure mean response times for Web pages with a

total size of less than 30 Kbytes (the mean bit rate of

the Web applications is rather small).

Secondly, only a greedy FTP connection is

combined with the VoIP traffic. This allows us to

evaluate the possible download rate for a bulk data

transfer. Different network delays of 1, 10 and

100 ms for the FTP connection shall yield further

insight into the performance of the investigated

approach.

The VoIP connection is modelled as bidirectional

flow with isochronous payload of 20 bytes every

20 ms (corresponding to G.729A). Including all

overheads on the ADSL link (RTP, UDP, IP,

PPPoE, AAL, ATM), this leads to an effective

amount of 159 bytes per voice packet. In the uplink

queue of the ADSL router, scheduling with absolute

priority of VoIP packets is assumed.

For this simulation, only packet-level performance

metrics have been used to yield an estimate of the

eventual speech quality. Recent investigations, e.g.

(Markopoulou, Tobagi, Karam, 2003), (James,

Chen, Garrison, 2004), find that the mean packet

loss ratio should not exceed 1%, and that packet

losses in clusters or bursts can be compensated to

different degrees by PLC (packet loss concealment)

algorithms of the voice decoder. According to this,

we assess the QoS of VoIP using the average packet

loss ratio as well as the distribution of loss burst

lengths. Two losses are considered to be part of the

same burst if less than 10 packets are transmitted

successfully in between. As the robustness of PLC

algorithms may vary, we plot the frequency of loss

bursts longer then 2, 5 and 8 packets in the Figures 4

and 5 below.

In the applied network model, all buffer sizes are

set to very large values. Thus, packet losses for the

VoIP connection only are caused by excessive delay

in the DSLAM queue and subsequent packet

dropping in the de-jittering buffer of the VoIP

phone. In this paper, we only consider a very simple

algorithm for the de-jittering buffer. All packets

arriving “in time” will be buffered till a threshold

time before being played out. Conversely, any

packets that arrive late will be dropped by the buffer.

The numerical results below assume a delay budget

of 50 ms between arrival at the DSLAM buffer and

packet delivery to the decoder.

All simulations have been performed for 100

sub-runs each of 10 minutes model time, the

confidence level is 95%. The simulations are

realized in ns-2 (http://www.isi.edu/nsnam/ns/).

3.2 Simulation II

Figure 3 illustrates the experimental structure. The

source signal holding about 20 seconds speech will

be coded and packtized into VoIP packets in the

VoIP sender. After transversing through the network

introduced in Simualtion I, the speech signal will be

recovered from those packets with the help of the

data extractor and the decoder. Finally, the PMOS

(PESQ MOS) value will be computed in PESQ by

comparing the degraded speech file with the original

one.

The increasing interest on applications like VoD

(Video on Demand) and interactive Internet gaming

leads to rising bandwidth demand. Correspondingly

the ISPs also offer higher bandwidth to their

customers. This higher bandwidth might also solve

the QoS problem of some applications having

comparatively low bandwidth requirements like

VoIP. In this simulation, besides the normal rate as

set in simulation I, doubled bandwidths are set to

both of ADSL up and down links for investigation.

The network model introduced in simulation I is

adopted. Two Web sessions running simultaneously

are set to the background load. The coding rate of

G.729 is set to 8Kbit/s which leads to 20 Bytes

payload for every 20 ms. We turn off the VAD

(Voice Activity Detection) function in codec to

maximize the bandwidth requirement of VoIP

application. For estimating voice quality, the mean

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272

Figure 3: Structure in simulation II.

packet loss ratio and the number of the loss burst in

20 s, as well as the mean PESQ value and the bad

PESQ ratio are used. The latter measure indicates

the relative frequency of PESQ values below 3.5. In

simulation I, the loss distance (the difference in

sequence numbers between two successively lost

packets) is defined as 10 under the assumption that

PLC is disabled in the decoder. According to (Levy,

Zlatokrilov, 2006), small values of loss distance can

be used when FEC (Forward Error Correction) or

PLC mechanisms are enabled. Based on the

assumption that the codec G.729 only can repair 30

to 40 ms losses (Perkins, Hodson, Hardman, 1998),

the number of loss bursts with length great than 2

(20 ms per packet) is depicted in Figure 6 below. In

both cases we assume that 100 ms delay budget

should be reserved for the uplink of the other side

and use a non-adaptive de-jittering buffer with size

of 150 ms in the receiver. The simulations have been

performed for 10x40 times; within each time the

whole transmission of the test speech file is

completed. In order to clearly separate to sub-runs, a

silence period of 300 ms is introduced between two

speech transmissions.

4 SIMULATION RESULTS AND

COMPARISONS

4.1 Simulation I

Figure 4 shows the VoIP quality in ADSL

downlinks with different delay budgets when

competing with 2 Web sessions. To ensure a VoIP

packet loss ratio below the tolerable value of 1%,

about 500 ms delay budget should be reserved for

the VoIP downlink (between DSLAM and the

receiver) which is far above the limitation.

Figure 5 shows the performance of VoIP and

TCP with packet dropping in ADSL router. As

discussed in Section 2, the introduced dropping ratio

should increase with the number of TCP

connections. In order to reduce the queueing delay

of VoIP packet in DSLAM to an acceptable level of

e.g. 50 ms, the queue length at arrival of a VoIP

packet should not exceed 4 packets (TCP packet

size: 1500 bytes, downlink bit rate: 1 Mbit/s). In the

case of only one connection, the TCP window size

therefore should be limited to 4 packets. With Web

sessions in the network, TCP packets from different

connections in a session can arrive simultaneously at

the DSLAM queue and provoke a larger queue

occupation. To ensure that the accumulated queue

length will not exceed 4, the window size for each

TCP connection should be further limited. Hence, a

very high drop ratio should be introduced which

may lead to low link utilization. High packet loss

ratios cause poor TCP behaviour during the start of

the connection and thus result in bad perceived

performance especially for Web applications.

Additionally, the dropping ratio should be adjusted

to a value that ensures the VoIP quality for the

lowest possible packet round-trip time (RTT) of all

TCP connections. In case of actually larger RTTs, a

certain fraction of the TCP packets will be in transit

on links or waiting in queues other than the

DSLAM. Thus, the TCP performance will be worse

than necessary.

As shown in Figure 5 (left side), discarding more

than 5% TCP packets can push the VoIP packet loss

ratio (fraction of late packets) below 1%, while the

frequency of packet loss bursts still can be serious.

We consider that the number of loss bursts with

length greater than 5 packets should not exceed 5

source

G.729

Packet

generator

Data

extractor

G.729

PESQ

degraded

signal

PMOS value

Network model

ns2 Model

IMPROVEMENT OF VOIP QUALITY BY PACKET DROPPING IN ADSL ROUTERS

273

within 10 minutes. To satisfy this requirement, it

follows from the figure that more than 10%

dropping ratio should be introduced. The right side

of Figure 5 demonstrates that then both FTP

throughput and mean Web response time deteriorate

considerably.

4.2 Simulation II

Figure 6 shows the VoIP performance in ADSL

downlinks when competing with 2 Web sessions.

In the case of 1 Mbit/s downlink bit rate, one

may find that our algorithm performances much

better on a short-time speech than on a long speech

(simulation I). The fraction of late VoIP packet can

be controlled below our criteria of 1% with a drop

period of 40 packets, while simulation I led to a

dropping ratio of more than 1/25. As shown in

Figure 6, the mean PESQ value can be quite good

with only 1/90 drop ratio. At the same time, the

mean number of loss bursts in 20 seconds is only

about 1. Nevertheless, more than 10% drop ratio is

required to hold the relative frequency of bad PESQ

values below 1%. This means that the user will not

more than once experience a bad VoIP quality

during 30 minutes.

Table 2 depicts the VoIP parameters with 2

MBit/s downlink bit rate on the ADSL link. Packet

dropping is not applied. The table shows that the

packet-level performance metrics do not indicate

QoS problems. The fraction of late VoIP packets in

20 s is only 1% which is almost within our criterion.

The mean number of loss bursts with burst length

greater than two is 0.4 times in 20 s. But the serious

situation of VoIP application under this scenario can

still be seen from the bad PMOS fraction which is

indicated as 15%. This means that the user will be

dissatisfied with the conversation quality more than

15 times within 33 minutes. The green curve in

Figure 7 addresses this problem more clearly.

According to the bursty background traffic, the VoIP

quality can be very different over time. In the total

100 simulation runs, there are 15 PESQ value drops

below 3.5, two estimates even fall below 3.0. The

worst PESQ value is indicated as 2.75 in Table 3.

Figure 6 indicates that dropping every 30

packet could be a good operation point in this

scenario. With this setting, only 0.06% of the VoIP

packets are lost, and the occurrence of bad PESQ

values can be almost suppressed (indicated to be

under 0.8%). However, the QoS can not be

guaranteed, since dropping of TCP packets can

trigger loss bursts of VoIP. When the lost TCP

segment is successfully retransmitted, the receiver

will send back a packet that acknowledges the lost

together with subsequent correctly transmitted

packets. This can lead to a sudden shift of the

transmission window and therefore cause a burst of

packets. This TCP burst can delay a sequence of

VoIP packets, leading to a VoIP loss burst and thus

degrading the voice quality. The sample curve of

PESQ values in Figure 7 highlights this aspect.

Although the PESQ estimate can be improved for

most of the speech intervals, the approach

sometimes can even make worst the speech quality.

This problem seems to be solved by further

increasing the TCP packet dropping ratio and

therefore limiting the TCP window size furtherly.

With a dropping ratio extended to e.g. 1/15,

sustained good PESQ values can be obtained,

varying from 3.71 to 3.75. As seen from simulation

I, this, however, will lead to severely reduced TCP

performance.

Figure 4: Performance of VoIP in ADSL downlinks in simulation I & Figure 7: PESQ values with different algorithms.

1e-06

1e-05

1e-04

0.001

0.01

0.1

0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1

Fraction of late VoIP packets

Number of loss bursts in 10 minutes

Delay budget / s

Late VoIP packets

Burst length > 2

Burst length > 5

Burst length > 8

2.5

3.5

5 10 15 20 25 30 35 40 45 50 55 60 65 70 75 80 85 90 95 100

PESQ MOS value

Number of iteration

without algorithm

Packet drop (1/30)

Packet drop (1/15)

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Figure 5: Performance of VoIP and TCP with packet dropping in simulation I.

Figure 6: Performance of VoIP with packet dropping in simulation II (single bandwidth).

Table 2: VoIP parameters with double bandwidth without packet dropping.

Fraction of VoIP packets Number of loss bursts (Burst length > 2) Bad PMOS (PMOS < 3.5)

1.01% 0.407 15.8%

Table 3: PESQ values from different algorithms (double bandwidth).

Mean Best Worst Bad fraction (<3.5)

No algorithm 3.654 3.754 2.750 0.15

Packet drop (1/30) 3.740 3.754 3.385 0.01

Packet drop (1/15) 3.754 3.754 3.711 0

5 CONCLUSION

In this paper we investigate the simple approach of

TCP packet dropping to reduce the queueing delay

of VoIP packets in the ADSL downlink under

different TCP traffic conditions. We assess the

performance of the algorithm both in a critical and a

more practical scenario. In the former case VoIP has

to compete for the limited bandwidth with TCP

traffics during a long time. In the latter one, a short-

time speech is investigated with single and double

bandwidth in the ADSL links. Our simulation results

indicate that VoIP suffers from QoS problems even

with high ADSL downlink bandwidth. In both cases,

the VoIP quality can be improved by TCP packet

dropping. The required high dropping ratio can,

1e-06

1e-05

1e-04

0.001

0.01

0.1

10 20 30 40 50 60 70 80 90

0.3

0.6

0.9

1.2

1.5

1.8

2.1

2.4

2.7

Fraction of late VoIP packets

Number of lossbursts in 20s

Drop period / packet

1Mbit/s: mean loss ratio

2Mbit/s: mean loss ratio

1Mbit/s: number of bursts (burst length > 2)

2Mbit/s: number of bursts (burst length > 2)

0.001

0.01

0.1

10 20 30 40 50 60 70 80 90

Bad PMOS fraction

PESQ MOS

Drop period / packet

1Mbit/s: mean PMOS

2Mbit/s: mean PMOS

1Mbit/s: mean bad PMOS fraction (PMOS < 3.5)

2Mbit/s: mean bad PMOS fraction (PMOS < 3.5)

0.001

0.01

0.1

10 15 20 25 30 35 40 45 50

Fraction of late VoIP packets

Number of loss bursts in 10 minutes

Drop period (in packets)

Fraction of later VoIP packets

Burst length > 2

Burst length > 5

Burst length > 8

0.2

0.4

0.6

0.8

10 15 20 25 30 35 40 45 50

FTP throughput / (MBit/s)

Web response time / s

Drop period (in packets)

FTP throughput 1

FTP throughput 10

FTP throughput 100

Web response time

IMPROVEMENT OF VOIP QUALITY BY PACKET DROPPING IN ADSL ROUTERS

275

however, lead to poor TCP behaviour, e.g. low FTP

throughput which yields low link utilization, and

long Web response time. Furthermore, the TCP

packet bursts indirectly caused by the dropping

algorithm can result in unstable VoIP quality.

ACKNOWLEDGEMENTS

This work was supported by the Sphairon Access

Systems GmbH.

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