Seminar Nasional Teknologi Informasi, Komunikasi dan Industri (SNTIKI) 5 ISSN : 2085-9902

Pekanbaru, 2 Oktober 2013

Analisa Quality of Service (QoS) IP Telephony dengan

metode Low Latency Queuing (LLQ)

Miftah Rahman Syahrial

Universitas Mercu Buana
Jl.Meruya Selatan, Kembangan, Jakarta Barat 11650. Telp: 021-5861779, 68640100, 5840816(hunting)
e-mail: miftahrahman86@gmail.com

Abstrak
Voice over IP pada era modern sekarang ini sudah sangat krusial. Teknologi ini bisa mengatasi
permasalahan yang muncul dalam telepon analog atau telepon tradisional ini adalah ketika pengguna,
terutama perusahaan, ingin melakukan komunikasi jarak jauh dari pusat ke kantor cabang dimana cost
alias biaya yang muncul ketika berkomunikasi dari pusat dan cabang berlangsung lama. Dalam
penerapannya, telekomunikasi yang menggunakan teknologi internet, mengalami beberapa hambatan
terutama dalam hal packet loss dan delay. Thesis ini membandingkan metode manajemen kongesti dalam
Quality of Service seperti Low latency queuing dengan metode-metode lainnya seperti First in First out
dan Class Based Weighted Fair Queuing, serta menganalisa efeknya terhadap penggunaan codec yang
sering dipakai seperti G.711 dan G.729.

Kata kunci: QoS, LLQ, delay, ITU-T G.114, codec

Abstract
Voice over IP in this modern era is almost crucial. This Technology can solve problem that arise
in analog telephony or traditional telephony is when user, especially in company, want to make
communication from main office to branch in a long distance where cost is highly sensitive (especially after
long period). In current deployment, telecommunication that using internet technology face some
obstacles, especially from bandwidth area. This thesis compares the method of Quality of Service
congestion management that called Low Latency Queuing with other methods such as First In First Out
and Class-Based Weighted Fair Queuing, and analyzing the effect on usual codec implementation like
G.711 and G.729

Keywords: QoS, LLQ, delay, ITU-T G.114, codec

1. Introduction
The survey results[1] on the use of ICT (Information & Communication Technology) in
the business sector explains that the business sector activity is most often done to send and
receive email at 97.69%, while the activity which has the lowest percentage is at 0.03% is hotel
promotion, while VoIP teleconference at 13.54%.

Graph 1. Survey results in the use of ICT in the business sector

Seminar Nasional Teknologi Informasi, Komunikasi dan Industri (SNTIKI) 5 ISSN : 2085-9902
Pekanbaru, 2 Oktober 2012

From the graph above shows that users have started to consider using VoIP in their
networks, not only because it can be integrated with existing data networks (computer
networks), VoIP is also emerging as a good solution because of the problems that arise in a
traditional analog phone or telephone, when users want to communicate from place to place
such as from a branch office to the head office will increase telecommunications costs. From
this problem, many companies are looking for alternative method to communicate, one of which
is by using internet technology, the Voice over IP (VoIP) phones that use the IP network as a
telecommunications medium.
In its application, IP Telephony (which is sometimes also called VoIP), encountered
some problems, especially in terms of bandwidth availability, packet loss, and delay. This thesis
focus on how to maximize the Quality of Service (QoS) for Internet-based telecommunications
by analyzing the delay caused by each of the existing congestion management methods,
especially LLQ which is the focus of this paper, based on existing recommendation that the
ideal delay is less than 150ms[2] and the the method effect if integrated with codecs that is
often used.
In the implementation of QoS, there are a variety of ways to maximize the performance
of existing networks; one of them is Congestion Management. Method of congestion
management that will be analyzed more deeply is LLQ. LLQ method is a method of combination
of Priority Queuing (PQ) and Class-based Weighted Fair Queuing (CBWFQ)[3]. This method will
also analyze the effectiveness from LLQ when integrating with codecs.
The codec used is G.711 mu-law codec (or often called a u-law) and codec G.729br8
(Annex B) which has a VAD (Voice Activity Detection) and CNG (Comfort Noise Generation).
Each codec have its algorithm, compression, and separate signaling technique[4] which led
delay that impacted the network performance[5].
This paper discusses the extent to which LLQ can suppress delay are examined for
comparison against other methods of congestion by measuring one-way trip delay and the
effectiveness of the codec G.711 and G.729.

2. Research Method
The research methodology used in this paper is prototyping simulation technique which
will be described as follows.

Cisco Router 2800 Series Cisco ASA Cisco Router 2800 Series
Firewall 5510

PC 1 PC 2

Cisco Catalyst 3750 PoE Cisco Catalyst 3750 PoE

Cisco IP Phone 7940 Cisco IP Phone 7940

Dial-peer 1234 Dial-peer 5678

Picture 1. Topology used in this paper

To simulate a congested network which causing delay, the router interface that leads to
another router, one of them (left router) should be made as low as possible in terms of transfer
rate (traffic shaping). This study uses the FastEthernet technology (100Mbps) which will be
reduced to 20kbps.
Seminar Nasional Teknologi Informasi, Komunikasi dan Industri (SNTIKI) 5 ISSN : 2085-9902
Pekanbaru, 2 Oktober 2012

Picture 2. Traffic shaping configuration on router

The results obtained are as follows:

Table 1. traffic shaping before and after

After one of the router interface transfer rate reduced, we conducted experiments
implementing each congestion management configuration from FIFO, CBWFQ, and LLQ (FIFO
is the default method of congestion management) one by one respectively on the left router and
doing data transfer activities in that congested network and at the same time trying to do the
telecommunication session from IP Phone to another for 10 attempts (and compare delay result
each attempt). The results are seen through the command "debug ip rtp protocol" in the router
command-line interface[6].

With FIFO method:

FIFO
no. R1 (Tx) R2 (Rx) Delay
1 37,562 39,363 1,801
2 4,2 5,711 1,511
3 44,323 46,611 2,288
4 12,15 17,564 5,414
5 45,512 54,788 9,276
6 1,456 3,945 2,489
7 13,444 18,782 5,338
8 12,561 15,777 3,216
9 9,12 10,566 1,446
10 10,444 11,62 1,176
Table 2. FIFO result after 10 attempts
Seminar Nasional Teknologi Informasi, Komunikasi dan Industri (SNTIKI) 5 ISSN : 2085-9902
Pekanbaru, 2 Oktober 2012

With CBWFQ method:

Picture 3. CBWFQ configuration on router

The results obtained are as follows:

CBWFQ
no. R1 (Tx) R2 (Rx) Delay
1 30,703 35,261 4,558
2 4,15 5,444 1,294
3 1,224 1,611 0,387
4 0,15 2,564 2,414
5 0,123 0,788 0,665
6 12,23 12,945 0,715
7 4,41 5,782 1,372
8 3,414 7,777 4,363
9 14,2 15,566 1,366
10 51,345 52,62 1,275
Table 3. CBFWQ result after 10 attempts
Seminar Nasional Teknologi Informasi, Komunikasi dan Industri (SNTIKI) 5 ISSN : 2085-9902
Pekanbaru, 2 Oktober 2012

With LLQ method:

Picture 4. LLQ configuration on router

The results obtained are as follows:

LLQ
no. R1 (Tx) R2 (Rx) Delay
1 12,316 12,338 0,022
2 14,553 14,574 0,021
3 30,685 30,706 0,021
4 21,457 21,478 0,021
5 4,004 4,146 0,142
6 1,122 1,123 0,001
7 45,114 45,224 0,11
8 55,48 55,488 0,008
9 13,489 13,589 0,1
10 13,567 13,633 0,066
Table 5. LLQ result after 10 attempts

After the results of each method are obtained, LLQ also tested the effectiveness for
codec G.711 and G.729

With G.711 codec:

Picture 5. G.711ulaw codec configuration on router

Seminar Nasional Teknologi Informasi, Komunikasi dan Industri (SNTIKI) 5 ISSN : 2085-9902
Pekanbaru, 2 Oktober 2012

With G.729 codec:

Picture 6. G.729br8 codec configuration on router

The comparison of each codec with LLQ implementation is as follows (calculated based
on the length of the telecommunications sessions per second):

Delay Comparison Table from each codec

with LLQ method
Detik G.711ulaw G.729br8
1 0,069 0.022
2 0,042 0.141
3 0,023 0.013
4 0,14 0.021
5 0.102 0.102
6 0,042 0.002
7 0,061 0.11
8 0,061 0.08
9 0,07 0.105
10 0,08 0.06
11 0.101 0.109
12 0.006 0.104
13 0.09 0.124
14 0.05 0.121
15 0.077 0.144
16 0.091 0.147
17 0.051 0.124
18 0.113 0.149
19 0.101 0.137
20 0.077 0.144
Table 6. Delay Comparison Table from each codec with LLQ method
Seminar Nasional Teknologi Informasi, Komunikasi dan Industri (SNTIKI) 5 ISSN : 2085-9902
Pekanbaru, 2 Oktober 2012

3. Result and Analysis

From the above experiments, it can be concluded that the LLQ is congestion
management method that is best to implement on a network that integrates voice in it, but this
method have not been able to reduce the overtime delay produced by the G.729 codec.

3.1. Comparative analysis of congestion management methods

From the experiment results obtained, both LLQ and CBWFQ can suppress one-way
trip delay of less than 150ms, but delay generated by CBWFQ tends to be unstable.

Graph 2. The comparison between congestion management methods

3.2. Codec comparison analysis

From the above table, although the time delay generated by G.729br8 codec can be
suppressed with LLQ method but the resulting delay codec G.729br8 still tends to increase with
the length of the session conversation, while the delay generated by codec G.711ulaw was less
affected by how long do sessions and tend to be stable.

0.4

0.35

0.3

0.25

0.2 G.729br8
G.711ulaw
0.15

0.1

0.05

0
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20

Graph 3. The overtime delay comparison between the codec using LLQ
Seminar Nasional Teknologi Informasi, Komunikasi dan Industri (SNTIKI) 5 ISSN : 2085-9902
Pekanbaru, 2 Oktober 2012

4. Conclusion
LLQ is the best voice-enabled network congestion management method for now, but
have not been able to help more in reducing overtime delay from G.729.
Further study is needed on how to improve the performance of LLQ to help reduce the
burden of routers when processing variety of codecs.

Reference:
[1] Hasil Survey Penggunaan Teknologi Informasi dan Komunikasi (TIK) di Sektor Bisnis Indonesia. Pusat
Data dan Sarana Informatika Kementrian Komunikasi dan Informatika Indonesia. Hal: 69. 2011
[2] ITU-T G.114. Series G:Transmission System and Media, Digital Systems and Networks. International
Telephone Connections and Circuits – General Recommendations on Transmission Quality for an Entire
International Telephone Connection. One-way Transmission Time;2003
[3] Martin J. Fisher, Denise M. Bevilacqua, John F. Shortle. Approximating Low Latency Queuing Buffer
Latency. IEEE Computer Society on the 4 th advanced international conference on Telecommunications.
Athens. 2008
[4] Kevin W. Implementing Cisco Unified Communications Voice over IP and QoS (CVoice) Foundation
Learning Guide. Fourth Edition. Cisco Press.2011:128-130
[5] Selvakumar V. Evaluating the Quality of Service in VoIP and Comparing Various Encoding
Techniques. MSc Thesis. UK:University of Bedfordshire;2011
[6] Implementing Cisco Quality of Service (QoS) - Student Guide. Cisco Press.2006