6th International Conference on Electrical and Computer Engineering

ICECE 2010, 18-20 December 2010, Dhaka, Bangladesh

QoS Optimization and Performance Analysis of NGN

Umma Hany1, A. B. M. Siddique Hossain2, Pran Kanai Saha3
1
The University of Asia Pacific, Dhaka, Bangladesh
ummahany@gmail.com
2
American International University of Bangladesh, Dhaka, Bangladesh
siddique@aiub.edu
3
Bangladesh University of Engineering and Technology, Dhaka, Bangladesh
sahapk@eee.buet.ac.bd

Abstract---- Next Generation Network (NGN) is a packet based NGN has been optimized by applying the developed
network which can support the expansion of broadband and technologies. Then the network is simulated and analyzed to
introduction of triple play (Voice +Data +Video) over fixed and determine the maximum capacity and QoS over NGN.
mobile line. Before deployment of NGN, the capacity and Quality
of Service (QoS) over NGN must be ensured. In this paper, the
technologies proposed by Huawei have been applied to ensure
II. SIMULATION AND RESULTS
the capacity and QoS over NGN. Then the optimized NGN is A. Simulation
simulated to observe the effect of the quality factors on the voice The QoS control technologies are applied on the following
performance. The simulation results are then analyzed and NGN network of Banglaphone as shown in Fig. 1. Then the
verified by theoretical analysis to determine the maximum voice over NGN is simulated using the “QoS Analyzer” tool
capacity and QoS which is possible to achieve over the proposed of Huawei. Five simulation agents have been set for five VoIP
NGN. The simulation results show a good agreement with that of
nodes. During simulation, two different codec is used and
the theoretical results.
incremented throughput test is done through generating traffic
Keywords: NGN, QoS, Voice on internet protocol. flow with changed packet size. The highest traffic flow is
carried out from 16:00:00 to 18:40:00.
I. INTRODUCTION
QoS can be ensured by reducing the link failures and the
deterioration of quality due to overload. In Next Generation IP
network, IP lacks the traffic engineering (TE) mechanisms
capable of offering a differentiated QoS, while efficiently
allocating the network resource. Factors affecting the QoS
over NGN are speech encoders, delay, jitter, packet loss and
echo. The capacity of NGN is highly dependent on the chosen
speech codec. There are abundant of activities in developing
protocols, speech encoders and optimization services to ensure
the capacity and QoS over NGN. ITU-T developed and
standardized a series of audio codec [1]-[3] ranging in bit rates
from 5.3-64 kbps. Multi-Protocol-Level-Switching (MPLS)
[4] offers a way of incorporating TE mechanisms into IP. The
integration of differentiated services (DiffServ) [5] with
MPLS guarantees the QoS for a broad range of multiservice
traffic. ITU-T defines the QoS control NGN architecture [6] in
which service-related functions is independent of the
transport-related technologies. Hence, the transport stratum is
Dotted lines indicate signaling path and solid lines indicate voice path.
responsible for admission and resource control based on the
network policy and the resource availability. The service Fig. 1 Next Generation Network topology with 5 VoIP nodes
control function (SCF) is responsible for the application
signaling for the service setup. The open issue of the ITU-T Hence, the Softswitch [7] in the call control layer provide
QoS control is that it is on per-call basis. Thus, the QoS guaranteed QoS and network efficiency with real-time QoS
control in the core network will be a burden. The reliability monitoring and dynamic flow control (using different codecs
and security in the core network is another issue. The QoS and rejecting high-bandwidth applications). The Universal
control mechanism can be simplified by using the Media Gateway (UMG) in the bearer control layer improves
performance monitoring information in the core network. the bandwidth utilization and reduces the effect of the quality
Huawei proposed a service-independent QoS control NGN factors by adopting Advanced Voice Quality Assurance
architecture [7] in which the network is separated in layers technologies [7]. It is achieved by adopting different
(service, call control, bearer control) and is based on a unique encoding/decoding, Voice Activity Detection (VAD),
packet switched core network for all types of access networks, Comfort Noise Generation (CNG), dynamic buffering, lost-
services and terminals. Here, QoS assurance technologies are packet compensation and echo cancellation technology.
incorporated in each layer to ensure the QoS [7], [8] enabling Session Border Controller (SBC) [8] at the edge of the IP
the network to provide efficiency, reliability and security in network is used to solve the IMS (IP Multimedia Subsystem)
the core network. In this paper, the capacity and QoS over technical problems as Network security, NAT traversal, QoS
assurance of packets and media streams.

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B. Results 3) Jitter
1) Packet Loss Ratio Fig. 6 and 7 show that jitter delay for G.729 is more than G.711.
Fig. 2 and 3 show that packet loss using G.729 is more than G.711.

Fig. 6 G.711 Jitter Time Graph (on Oct 29, 2008 3:10 PM-10:10 PM)
Fig. 2 G.711 Loss Time Graph (on Oct 29, 2008 3:10 PM-10:10 PM)

Fig. 3 G.729 Loss Time Graph (on Oct 30, 2008 2:00 PM to 10:00 PM) Fig. 7 G.729 Jitter Time Graph (on Oct 30, 2008 2:00 PM to 10:00 PM)
2) Delay 4) MOS
Fig. 4 and 5 show that delay for G.729 is more than G.711.
Fig. 8 and 9 show that MOS using G.711 is better than G.729.

Fig. 4 G.711 Delay Time Graph (on Oct 29, 2008 3:10 PM-10:10 PM) Fig. 8 G.711 MOS Time Graph (on Oct 29, 2008 3:10 PM-10:10 PM)

Fig. 5 G.729 Delay Time Graph (on Oct 30, 2008 2:00 PM to 10:00 PM)
Fig. 9 G.729 MOS Time Graph (on Oct 30, 2008 2:00 PM to 10:00 PM)

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5) Throughput R = 93.2-Id-Ief (4)
Fig. 10 and 11 show that the throughput using G.711 is greater than Mean opinion score (MOS) [11] is to evaluate the voice
G.729 quality according to the scoring standards of ITU-T. MOS is
calculated from R as follows. Table 3 illustrates the ITU-T
voice quality at different network conditions.
MOS = 1 < 1 + (0.035R) + (R(R – 60) (100 – R) 7.0e-06) < 4.5 (5)
TABLE III ITU-T STANDARD VOICE QUALITY OVER NGN
Parameters and
Good Poor Bad
Service
ITU-T MOS 4.0-5 3.5-4 3-3.5 1.5-3 0-1.5
Delay ≤40ms ≤100ms ≤400ms
Standard Loss ≤0.1% ≤1% ≤5%
Jitter ≤10ms ≤20ms ≤60ms
G.711 Excellent Good Fair
Voice G.729 Good Good Poor
Fig. 10 G.711 Throughput Time Graph G.723.1 Good Almost Good Fair

2) Simulation results analysis

MOS is calculated by replacing the simulation results (delay,
jitter and packet loss) into equation (1)-(5) and compared to
the ITU-T standard to evaluate the voice quality as in Table 4.
TABLE IV SIMULATION RESULTS ANALYSIS TO EVALUATE QoS
E-Model
Agent 3 – Agent 4 at
packet loss between Simulation results
analysis ITU-T
Considering the

standard

Packet
Voice

Value
17:35

Delay

Jittter

MOS
Loss
(ms)

(ms)

(%)
R-
Quality

Fig. 11 G.729 Throughput Time Graph G.711 6.8 0.3 0 93 4.4 Excellent
G.729 12 0.25 0.17 81 4 Good
III. RESULTS ANALYSIS AND VERIFICATION
aggregated
simulation

G.711 6.5 0.3 0 93 4.4 Excellent

results

A. QoS Analysis
Using

1) Theoretical Analysis G.729 7.5 0.3 0.09 81.6 4 Good

Factors affecting the QoS are encoding mode, network Delay
(ms), Jitter, Packet loss, packet doubling and echo. Different
parameters and equipment impairment factors of the VoIP The simulation result analysis shows that due to the QoS
codecs are illustrated in Table 1. The Table 2 presents the optimization, the effect of the voice quality factors are
equipment impairment factor considering packet loss which is reduced and NGN provides good voice quality which is not
defined as follows: significantly degraded due to the utilization of different
codecs.
Ief = Ie + 30ln(1+ 15e) (1)
Ief: it is related to packet loss, e= packet loss ratio B. Capacity Analysis
TABLE I AUDIO CODEC PARAMETERS OF VoIP CODECS [1]-[3],[9] 1) Theoretical analysis
Bit Framing Equipment Each VoIP packet includes the headers at the various protocol
Para- Payload Packets
rate interval Impairment layers such RTP 12 bytes, UDP 8 bytes, IP 20 bytes, Ethernet
meters (Bytes) /s, Np
(Kbps) (ms) Factor, Ie 26 bytes and the payload comprising the encoded speech for a
G.711 64 20 160 50 0 certain duration depends on the codec deployed.
G. 723.1 6.3 30 24 33 15
G.729 8 20 20 50 10 or 11 OHhdr = HRTP+ HUDP+ HIP +HMAC (6)
Packet length = OHhdr + payload (7)
TABLE II EQIPMENT IMPAIRMENT FACTORS FOR DIFFERENT Payload = number of payload bits/s ¯ Framing Interval (s) (8)
CODEC CONSIDERING PACKET LOSS [9] Bandwidth = packet length ¯ number of packets per sec (9)
%packet loss 0 0.5 1 1.5 2 3 4 8 16 Let n be the maximum number of sessions which is supported
G.729a 11 13 15 17 19 23 26 36 49 by NGN. n is defined as follows:
G.723.1a 15 17 19 22 24 27 32 41 55 Data Rate
n= (10)
The delay impairment factor can be defined as follows: Bandwidth Occupied

Id= 0.024d + 0.11 (d-177.3) H (d-177.3) (2) Using equation (6)-(9) we can calculate the bandwidth
Id: it is related to end to end delay occupied by various Kinds of Voice Encoding/Decoding as
d=one-way delay (coding + network + de-jitter delay) [ms] illustrated in Table 5.
H(x) =0 for x<0 H(x) =1 for x > 0 TABLE V BANDWIDTH OCCUPIED USING DIFFERENT CODEC
The E-model [10] calculates the R from the network QoS Parameters G.711 G. 723.1 G.729
factors. The rating R is computed as follows: Framing interval (ms) 20 30 20
R = R0 – Icodec - Idelay - Ipdv – Ipacketloss (3) Bandwidth occupied (kbps) 89.78 22.49 33.78

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The capacity using different codec has been calculated using The call connected ratio is defined as follows:
equation (10) which has been plotted in Fig. 12. Fig. 12 shows Number of call attempts
that maximum capacity can be achieved using G.723.1 codec. Call Connected Ratio =
Number of call connected
X 100 (12)
Thus NGN provides high capacity using different codec. Fig.
13 shows the comparison of TDM and NGN capacity. Comparing the plots in Fig. 14 and 15, it is observed that at
the peak hour call connected ratio for DTCL is 59.5%-65%
and the call connected ratio for Teletalk is 96.5%-98.5%. Thus
Call connected ratio of IP based NGN is far better than TDM
based circuit switched network.

V. CONCLUSION
The Optimized NGN improves the bandwidth utilization and
Bit Rate
reduces the affect of the quality factors using the developed
Fig. 12 Maximum simultaneous VoIP nodes technologies. Analyzing the simulation results for G.711 and
G.729 codec, it is observed that better capacity is achieved
using G.729. Thus, optimized NGN is able to provide high
VoIP capacity using different codecs (ranging in bit rates from
5.3-64 kbps) as per the capacity requirement. The result
analysis also shows that excellent voice quality is achieved
using G.711 codec where good voice quality along with high
capacity is achieved using G.729 codec. Thus optimized NGN
is able to provide high voice quality which is not significantly
degraded due to the chosen codec and other quality factors
Fig. 13 Comparison of TDM and VoIP capacity for 2 Mbps data rate (delay, jitter and packet loss). Thus it is observed by analyzing
the results that high VoIP capacity and good voice quality is
2) Simulation (Throughput) analysis
possible to achieve over the optimized NGN. For the new
Throughput is the amount of data in bits that is transmitted services as fixed mobile convergence and IPTV, QoS control
over the channel per unit time. VoIP capacity can be for mobility and the multicast condition must be developed.
calculated by replacing the maximum throughput (simulation
result for 1 Mbps data channel) into equation (10) as follows:
Acknowledgement: Authors would like to acknowledge for using
Maximum Throughput the resources and support provided by Huawei Technologies Ltd,
VoIP Capacity, n = (11)
Bandwidth Occupied Banglaphone Ltd, Dhaka Telecom Co. Ltd, Teletalk Bangladesh Ltd.
The simulation results show that using G.711, the average and the department of EEE, BUET.
throughput at the peak hours is 0.92 and using G.729 it is
0.856 Mbps. Replacing the average throughput and the
REFERENCES
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