Improving QoS of IPTV and VoIP Over IEEE 802.11n

Computers and Electrical Engineering xxx (2014) xxx–xxx
Contents lists available at ScienceDirect
Computers and Electrical Engineering

journal homepage: www.elsevier.com/locate/compeleceng
Improving QoS of IPTV and VoIP over IEEE 802.11n q

Saad Saleh a,⇑, Zawar Shah b, Adeel Baig a,c
a
School of Electrical Engineering and Computer Science (SEECS), National University of Sciences and Technology (NUST), Islamabad, Pakistan
b
Whitireia Community Polytechnic, Auckland, New Zealand
c
College of Computer and Information Systems, Al Yamamah University, Saudi Arabia
a r t i c l e i n f o a b s t r a c t
Article history: Tremendous growth rates of Internet Protocol Television (IPTV) and Voice over Internet
Received 15 May 2014 Protocol (VoIP) have demanded the shift of paradigm from wired to wireless applications.
Received in revised form 23 October 2014 Increased packet loss with continuously varying wireless conditions make the transmission
Accepted 23 October 2014
a challenging task in wireless environment. Our study investigates and proposes improve-
Available online xxxx
ment in the transmission of combined IPTV and VoIP over the IEEE 802.11n WLAN. Our
major contributions include the analytical and experimental investigations of (1) transport
Keywords:
layer protocol UDP/TFRC for IPTV and VoIP, (2) optimal physical layer parameters for IPTV
IPTV
VoIP
and VoIP, (3) proposition of wireless enhancement of TFMCC (W-TFMCC) to enhance the
DCCP capacity and Quality of Service (QoS) of wireless IPTV and VoIP. Analytical and experimen-
TFRC tal evaluations show a 25% increase in capacity using TFRC with 167% more bandwidth
Multi-casting share to TCP. Our study shows that use of W-TFMCC with optimal parameters can enhance
TFMCC IPTV and VoIP capacity by 44%.
Ó 2014 Elsevier Ltd. All rights reserved.
1. Introduction
Internet Protocol Television (IPTV) is one of the fastest growing applications which has gained huge growth rates in the
past few years. Number of IPTV users are expected to increase by 500% from 2011 to 2016 [1]. Large growth rate with
increased user’s interest motivate us to study transmission of IPTV with an aim to provide better Quality of Service (QoS).
IPTV offers a number of advantages over its predecessor analog technologies. Major advantages of IPTV include user inter-
action, video on demand service, economic and better Quality of Service (QoS). Architecture of IPTV includes three entities:
video head end, transport network and video receiver. Video head end is placed at the server side and it has the tasks of video
encoding and transmission of video and audio to the user end. Transport network is the entity which plays the most crucial
rule because it incorporates jitter, delay, scrambling and packet loss effects during the transmission of video. Transport
network includes both wired and wireless medium. Inside the transport network, a number of queues having the parallel
storing capabilities which shuffle the packets. Video receiver is the last entity which has the task of decoding information,
eliminating delay and jitter factors and managing a reliable QoS at the user end.
Voice over Internet Protocol (VoIP) is another fastest growing internet application which has obtained huge growth rates
in the past few years. There are 10 times more VoIP users than IPTV users [1]. Major factors for VoIP success are cheap calling
rates, better QoS and better penetration among end users. VoIP uses bi-directional traffic and has more challenging
q
Reviews processed and recommended for publication to the Editor-in-Chief by Associate Editor Dr. Ziya Arnavut.
⇑ Corresponding author.
E-mail addresses: saad.saleh@seecs.edu.pk (S. Saleh), zawar.shah@whitireia.ac.nz (Z. Shah), adeel.baig@seecs.edu.pk (A. Baig).
http://dx.doi.org/10.1016/j.compeleceng.2014.10.017
0045-7906/Ó 2014 Elsevier Ltd. All rights reserved.
Please cite this article in press as: Saleh S et al. Improving QoS of IPTV and VoIP over IEEE 802.11n. Comput Electr Eng (2014), http://
dx.doi.org/10.1016/j.compeleceng.2014.10.017
2 S. Saleh et al. / Computers and Electrical Engineering xxx (2014) xxx–xxx
requirements for packet loss and delay than IPTV. Transmission of VoIP requires limited packet loss and delay for all users
which becomes challenging when optimum route changes for all users.
Currently, wired access links are preferred by service providers for transmission of IPTV and VoIP services owing to min-
imum packet loss and delay in the wired links. Transmission of IPTV and VoIP becomes challenging in the wireless environ-
ment because major bandwidth restriction occurs at the user end having wireless Access Point (AP) [2]. Packets drop from
queues of the wireless AP which make it difficult to meet the QoS constraints of IPTV and VoIP. Moreover, range and data rate
are also limited in wireless links owing to the continuously varying wireless conditions. Users demand, ease of access and
freedom of mobility require an insight investigation for transmission of IPTV and VoIP over wireless networks.
IEEE 802.11n Wireless Local Area Network (WLAN) is the latest standard proposing data rates upto 600 Mbps (theoret-
ically) and 300 Mbps (practically). IEEE 802.11n is equipped with a number of features which include its Multiple Input
Multiple Output (MIMO) technology and frame aggregation mechanisms at Medium Access Control (MAC) layer and Physical
(PHY) layer. Frame aggregation combines multiple frames at MAC layer and PHY layer level. Major advantage of frame aggre-
gation includes the reduction of header over-head time and also the reduction in the collision time. Our previous study for
IPTV and VoIP capacity over IEEE 802.11n shows that aggregation of 4 packets is the optimal aggregation size for capacity
enhancement of IPTV and VoIP [3,4].
Enhancing QoS of IPTV and VoIP with increased capacity over IEEE 802.11n has been actively discussed in various studies
due to the challenging IPTV constraints over WiFi. Packet loss is a major factor which results in capacity reduction for IPTV
and VoIP. IPTV and VoIP are extremely susceptible to packet loss because both use User Datagram Protocol (UDP) at the
transport layer.
UDP is a constant bit rate protocol. By use of UDP, packets accumulate at the AP which results in congestion at the AP. All
packets crossing the limits of queue size are dropped at the AP. Our study shows that UDP provides less delay but it increases
packet loss which becomes the bottleneck for other users. Our previous study [3,4] shows that Datagram Congestion Control
Protocol (DCCP) is the better suited protocol for transmission of IPTV and VoIP. DCCP has two variants namely TCP-like and
TCP Friendly Rate Control (TFRC). TCP-like offers high reliability and decreases its data rate much more rapidly than TFRC.
This makes it suitable for all applications demanding less packet loss. On contrary, TFRC offers a nearly constant data rate by
maintaining its data rate according to varying conditions of the network. Behaviour of TFRC makes it suitable for all appli-
cations which require less delay. Our investigations [3,4] reveal through simulations that TFRC gives better performance than
UDP for transmission of IPTV and VoIP over IEEE 802.11n.
In this paper, we aim to develop an analytical model for transmission of IPTV and VoIP over IEEE 802.11n.1 Transport layer
protocols UDP and TFRC are modelled by their behaviour in the wireless environment. Extensive experiments are performed to
validate the analytical results. Various physical layer parameters are modelled through SIFS, DIFS and default behaviour of wire-
less environment. Optimal values of queue size, contention window, SIFS and DIFS are proposed. Analytical values of physical
layer parameters are compared with experimental results. We propose Wireless TCP Friendly Multicast Congestion Control Pro-
tocol (W-TFMCC). TFRC and TCP-like suffer low capacity because both use the unicast mechanisms. Capacity can be enhanced
significantly by shifting unicast transmissions to multicast transmissions. In this paper we present the results of TCP Friendly
Multicast Congestion Control Protocol (TFMCC). TFMCC is designed for wired networks which suffer low packet loss and all
users are nearly in the same conditions. TFMCC keeps a track of the user facing worst packet loss conditions and adjusts its
sending rate according to the worst case user. This is highly unsuitable for the wireless medium because all users are present
in different environments. TFMCC forms channel groups based only upon users demands. Transmission for the worst case user
would lead to lower data rate even if a single user is having high packet loss rates or Round Trip Times (RTT). We suggest a
group based protocol which keeps a track of the various conditions experienced by different users. Our study shows that per-
formance of W-TFMCC is greater than UDP/TFRC/TFMCC if at least two or more users are watching same channels. Performance
of W-TFMCC is equal to TFRC/TFMCC when all users are watching different channels.
Our contributions in this work are (i) analytical and experimental evaluation of transport layer protocols UDP/TFRC for
transmission of combined IPTV and VoIP over IEEE 802.11n, (ii) analytical and experimental investigation of optimum phys-
ical layer parameters of combined IPTV and VoIP over IEEE 802.11n, (iii) proposition of a new group-based multicast protocol
W-TFMCC with performance analysis over UDP/TFRC/TFMCC through simulations and experiments.
The rest of the paper is organized as follows. Section 2 presents the related work for IPTV and VoIP over WLANs. Section 3
presents the experimental scenario and data rate estimation for IPTV and VoIP. Section 4 presents the IPTV and VoIP capacity
analysis over UDP and TFRC along with fairness analysis with TCP traffic. In Section 5, we show the optimal values of IEEE
802.11n analytically and experimentally. Section 6 presents the performance improvement using multicast mechanism and
our proposed W-TFMCC protocol. Section 7 presents the comparison of our results with previous state of the art approaches.
Section 8 concludes the paper.
1
Initial results of this research appeared in
Saad Saleh, Zawar Shah, Adeel Baig, ‘‘Capacity Analysis of Combined IPTV and VoIP Over IEEE 802.11n’’, In the IEEE Conference on Local Computer
Networks (LCN), Sydney, Australia, October 2013.
Saad Saleh, Zawar Shah, Adeel Baig, ‘‘IPTV Capacity Analysis using DCCP over IEEE 802.11n’’, In the IEEE proceedings of Vehicular Technology Confer-
ence (VTC), Las Vegas, USA, September 2013.
S. Saleh et al. / Computers and Electrical Engineering xxx (2014) xxx–xxx 3
2. Related work
Transmission of IPTV over Wireless Local Area Networks (WLANs) is a challenging task because of large packet loss, large
delay and minimum available bandwidth. A number of investigations have been made for availability of wireless IPTV. In [5],
authors evaluate the capacity trends of IPTV over IEEE 802.11b and IEEE 802.11g networks. They conclude that IPTV users
and WLAN data rate have non-linear relationship with each other. Their findings show that IEEE 802.11b and IEEE
802.11g networks can support 2 and 6 IPTV streams respectively. In [6], authors apply the fluid model flow analysis to deter-
mine IPTV capacity. They incorporate buffer size, network hops and show a non-linear relationship between them for reliable
QoS of IPTV. In [7], an experimental investigation has been made to evaluate the capacity of IPTV over IEEE 802.11n WLAN.
Their QoS findings show that outdoor environment deteriorates IPTV performance significantly while indoor environment
can support dozens of low resolution users with reliable QoS. In [8], Kilik and Amadou implemented a practical test bed
of IPTV to observe the behaviour of various users in IPTV channel streaming. However, their research is not focused over
the improvement in various layers of IPTV but is limited to performance of current IPTV architecture. In another study
[9], Piamrat et al. study the transmission of IPTV over the wireless home network. Authors analyze the performance of IPTV
over UDP and TCP transport layer protocols as well as various MAC layer protocols. Authors propose a solution of combined
usage of TCP with a coordinated link layer protocol. In [10], Chaparro et al. evaluate the bandwidth requirements for trans-
mission of high quality television content. Authors show that granularity of the estimation can be utilized efficiently for con-
tent generator to react to changes in utilization of the network.
VoIP has been a major focus of numerous studies owing to its high demand. In [11], authors evaluate the capacity of VoIP
using various codec and packetization intervals over IEEE 802.11b network. They show through simulations that IEEE
802.11b WLAN can support 3–11 users depending upon the wireless channel loss conditions. In [12], authors evaluate
the capacity of VoIP along with tracking capacity over IEEE 802.11b/g WLAN. They show through simulations and experi-
ments that combined VoIP and tracking capacity is 30% less than VoIP only capacity at higher packetization intervals. In
[13], authors show through experimental setup that IEEE 802.11b WLAN can support 15 calls having a packetization interval
of 20 ms. They prove their experimental results through simulations and analytical work. Transmission of combined IPTV
and VoIP introduces more challenges by incorporating different delays and packet loss thresholds for different devices. In
[14], authors study the transmission of combined IPTV and VoIP over IEEE 802.11n with varying number of hops. Authors
have shown that it is possible to run 3 IPTV streams along with VoIP connected for 2 hops only. They prove their findings
through simulations and experiments and prove that hop count is inversely related to capacity of IPTV and VoIP. Authors
conclude that performance of IPTV and VoIP is limited in IEEE 802.11b/g networks due to less throughput. Performance is
expected to enhance in IEEE 802.11n WLAN due to provision of high data rates.
Analytical model of Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA) has been developed in [15]. Author
develops a markov model to estimate the packet loss probabilities and estimates the throughput of IEEE 802.11b WLAN.
Performance of IEEE 802.11n has been evaluated in [16]. Authors evaluate the MAC and PHY layer mechanisms of VoIP
and show that performance of IEEE 802.11n for VoIP is significantly enhanced. Frame aggregation mechanism of IEEE
802.11n has been evaluated in [17]. Authors show that IEEE 802.11n can enhance channel utilization upto 95% for UDP traffic
by using frame aggregation mechanisms. They conclude that MAC level aggregation is less effective than PHY layer aggre-
gation. In [18], authors study the IEEE 802.11n mechanisms in comparison to the legacy frame protection mechanisms. They
show that IEEE 802.11n has enhanced QoS as compared to its predecessor standards.
Performance of DCCP for delay sensitive applications has been evaluated in [12]. Authors show that DCCP adjusts its data
rate continuously which results in a decrease in packet loss at the queues of AP. They show through simulations that DCCP
gives much more fair share in bandwidth to TCP traffic than UDP. Our previous studies [3,4] show through simulations that
TFRC can increase network capacity for IPTV and VoIP. Performance of DCCP is limited because it provides unicast mecha-
nisms. A number of investigations have been made to increase efficiency by proposing multicast mechanisms. In [19],
authors propose a DCCP enhancement, called Multi(Uni) DCCP, which transmits its streams based upon the number of
receivers in the network. Authors show that their protocol can not only provide an increase in capacity but also decrease
network congestion. In [20], authors propose an improvement to currently developed protocol TFMCC by changing the data
rate equations. Authors conclude that incorporation of the uni-directional delay improves the performance of TFMCC. In [21],
authors develop a framework for multicast transmission of multimedia services through wireless networks. Authors show
that their framework increases efficiency with better QoS as compared to TFRC.
Comparison of various studies [5–21] show that capacity of IPTV and VoIP is limited over WLANs due to low data
rates of IEEE 802.11b/g. High data rates of IEEE 802.11n motivate the concept of wireless IPTV and VoIP over IEEE
802.11n. To the best of our knowledge, very limited studies exist on the performance of IPTV and VoIP over IEEE
802.11n.
2.1. Transport layer protocols
A number of transport layer protocols exist which vary in their performance. Before probing into the performance of IPTV
and VoIP, we define various transport layer protocols used in this study.
2.1.1. User Datagram Protocol (UDP)

UDP provides a constant data rate with no handshaking dialogues and no guarantee of service, ordering or duplicate pro-
tection. UDP has no congestion control mechanism with extensive voice and video applications along with the use in Domain
Name System (DNS) and Routing Information Protocol (RIP).
2.1.2. Datagram Congestion Control Protocol (DCCP)

DCCP implements reliable connection setup, congestion control, explicit congestion notification, feature negotiation and
tear down. Provides flow based semantics similar to TCP but does not provide reliable in-order delivery. DCCP has two vari-
ants TCP-like and TFRC. Applications of DCCP include internet telephony, online multiplayer games and streaming media.
2.1.3. TCP Friendly Rate Control (TFRC) protocol

TFRC provides a congestion control mechanism for unicast flows operating in the internet and competing fairly with the
TCP traffic. TFRC varies its data rate continuously based upon the network congestion, packet loss rate and round trip time.
Internet telephony and streaming media are the popular applications of TFRC.
2.1.4. TCP Friendly Multicast Congestion Control (TFMCC) protocol

TFMCC provides an equation-based congestion control mechanism for multicast connections by extending the TFRC pro-
tocol from unicast to multicast domain. TFMCC is most suitable for multicast applications demanding a smooth rate includ-
ing streaming media based applications.
3. Modelling of IPTV and VoIP over IEEE 802.11n
This section presents the modelling of IPTV and VoIP over IEEE 802.11n. Experimental setup for transmission of IPTV and
VoIP is discussed. Bandwidth estimations for IPTV and VoIP are also presented for simulations and analytical evaluations.
3.1. Network scenario
We develop an analytical model and make experimental setup for performance of IPTV and VoIP over IEEE 802.11n. The
analytical model assumes N nodes communicating with each other inside the wireless AP range as shown in Fig. 1. Model
assumes that network nodes are lying inside the access point coverage area and all nodes get small packet loss and RTT from
AP. AP is connected to the internet which joins AP to the IPTV and VoIP servers present on the opposite side of the internet.
Matlab has been used to generate results of analytical model. All simulations referenced from our previous study have been
cited. For simulations, ns2 has been used as the simulation tool.2 Support of our patch [4] has been extended for simulations of
IPTV and VoIP over IEEE 802.11n.
For experimental analysis, we setup an experimental test-bed. Distributed Internet Traffic Generator (DITG) ver-2.8.1 is
used to generate IPTV, VoIP and FTP packets from the application layer [22]. Sender and receiver devices have DITG installed.
Packet loss readings, RTT and delays are collected from the receiver devices. Data rate of DITG is adjusted for the data rates of
High Definition Television (HDTV) and Standard Definition Television (SDTV) streams of IPTV. DITG can be tuned for DCCP
and UDP streams. Results of DITG were validated with results of previous studies [11]. IEEE 802.11n AP used for experiments
has model no. AN1020 25 and name ‘‘ADSL Wireless Modem’’. Transmitter and receiver devices are connected to each
other through the wireless AP. The experimental setup developed is shown in Fig. 1. Various parameters, adopted from
[16] used in simulations and experiments are shown in Table 1.
3.2. Data rate estimation of IPTV and VoIP
IPTV requires a picture resolution which takes into account a number of factors including the pixel quality given by lumi-
nance and chrominance. Luminance is the light intensity and chrominance is the colour depth. Moreover, a moving picture is
composed of a number of frames which move in series to make a moving picture. We take into account the frames per sec-
ond effect for IPTV bandwidth estimation. Another important factor is the size of the video resolution. Although a number of
resolutions exist but there are two standard sizes used namely SDTV and HDTV using 16:9 or 4:3 resolutions.
Compression schemes play an important role in estimating the amount of data to be transmitted through the network.
Moving Picture Expert Group (MPEG) has suggested a number of compression schemes and the most popular schemes
are MPEG-2 and MPEG-4. Studies have revealed that H.264 (MPEG-4) gives a much better compression ratio than the cur-
rently used MPEG-2 standard [23]. Data rate requirement plays the most important role in evaluating the capacity of IPTV for
deploying in a given network. We evaluate the data rate by taking into account all the factors. The various factors used in the
data rate calculation are shown in Table 2. Using all these factors the data rate required in uncompressed form is given by Eq.
(1).
D ¼ RH RV CF ð1Þ
2
S. McCanne and S. Floyd. ns Network Simulator. http://www.isi.edu/nsnam/ns.
Fig. 1. Network scenario for IPTV and VoIP.
Table 1
IEEE 802.11n access point parameters.
Parameter Value Parameter Value

DIFS 34 ls SIFS 16 ls
Slot time 9 ls Physical header 20 ls
Contention window (min) 15 TXOP limit 5
Channel bandwidth 40 MHz Bit error rate 0.000008
Table 2
Data rate requirement for various compression and resolution schemes.
TV F (fps) C Resolution RH RV Compression scheme Required rate (Mbps)

SDTV 24 2 640 480 MPEG-2 3.93
SDTV 24 2 640 480 MPEG-4 2.36
HDTV 24 3 1920 1080 MPEG-2 26
HDTV 24 3 1920 1080 MPEG-4 15.92
Here RH is the horizontal resolution and RV is the vertical resolution for the picture resolution. C is the chrominance factor
and F is the intensity of frames per sec used for the pictures.
The data rate obtained without any compression scheme is high (797 Mbps) which is not achievable for wireless environ-
ment. Performance of various compression schemes specially MPEG-2 and MPEG-4 motivates the use of compression to raw
data in order to decrease the data rate. Compression ratios of two popular schemes are given as follows [23].
MPEG-2 and H.263 Compression ratio (Hcomp ) = 30:1.
MPEG-4 and H.264 Compression ratio (Hcomp ) = 50:1.
A group of pictures N gop is generated which is arranged in a certain priority given by I; P and B frames. The data rate equa-
tion after applying compression schemes is shown in Eq. (2).
D ¼ RH RV CFN gop =Hcomp ð2Þ
Table 2 presents the data rates required after applying compression schemes. It is worth mentioning that our data rate
calculation is with the latest standards (resolutions, compression schemes and frames per seconds) which is in accordance
with the previous researches [5–7,9]. We used the standard IPTV packet size of 1366 bytes which contains 1288 bytes of pay-
load data [7]. On contrary to IPTV, VoIP requires less data rate. Data rate of VoIP depends upon the packetization interval,
amount of payload data and the transmission schemes G.711 and G.729.3 For our study, we use the popular VoIP codec
G.711 (64 kbps) with a 10 ms codec sample interval.
IPTV requires a one-way delay constraint which is 50 ms [24]. On contrary, VoIP requires two-way delay constraints
which must be less than 150 ms for both directions [12]. IPTV cannot tolerate a packet loss greater than 1% while VoIP allows
a packet loss upto 2% [24,12].
4. Optimal transport layer protocol for IPTV and VoIP
In this section, we investigate the optimal transport layer protocol for transmission of combined IPTV and VoIP over IEEE
802.11n. Only simulations were performed in our previous studies [3,4]. In this section, we extend our analysis by conduct-
ing extensive experiments and analytical evaluations. Firstly, we investigate IPTV and VoIP with UDP and TFRC for both
applications. Secondly, we study the performance of TFRC based applications in presence of UDP based applications. Lastly,
we evaluate the performance of TFRC based applications in presence of non-real time TCP traffic.
4.1. IPTV and VoIP capacity analysis over UDP and TFRC
In this subsection, we present the analytical framework for transmission of IPTV and VoIP over IEEE 802.11n. Our analyt-
ical model estimates the packet loss probabilities, queue utilization ratio and expected CSMA/CA wait-off time to estimate
the throughput of IPTV and VoIP over IEEE 802.11n. As layers of computer network protocols are independent of each other,
so analytical modelling of physical layer does not change by incorporation of any other protocol at transport layer of IPTV or
VoIP. Our aim is to investigate the capacity of IPTV and VoIP over IEEE 802.11n using UDP and TFRC at transport layer.
Let traffic arrival rate be defined by Ai at any time instant i and frame service rate at queue of access point be defined by Si
at time instant i. Based upon the frame service rate and frame arrival rate, queue utilization ratio Q i is given by following Eq.
(3).
Ai
Qi ¼ ð3Þ
Si
Based upon Eq. (3), average queue utilization ratio Q for a period of T time units is given by the average of summation of
queue sizes on every instant as shown in Eq. (4).
PT
Qi
Q ¼ Pi¼1
T
ð4Þ
i¼1 1
To determine the transmission rate, we evaluate the probability of successful transmission by station i. Let pi be the prob-
ability of successful transmission and sj be the transmission probability of station i. Station i transmits successfully if no
other station is transmitting. Eq. (5) shows the probability of successful transmission.
Y
N1
pi ¼ ð1 sj Þ ð5Þ
j¼0
j–i
If station i is transmitting, a collision occurs if at least one of the remaining stations transmits. Let ci be the collision prob-
ability of station i. Conditional collision probability is given by Eq. (6).
Y
N1
ci ¼ 1 ð1 sj Þ ð6Þ
j¼0
j–i
ci ¼ 1 pi ð7Þ
3
Cisco, ‘‘Voice Over IP – Per Call Bandwidth Consumption’’, available at http://www.cisco.com.
IEEE 802.11n uses Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA). Let W i be the expected wait-off
time for any station i with CW j as contention window size and pi as the collision probability with Ai as expected number
of transmission attempts. Let m be the retry limit. Average wait-off time E½W i and average transmission attempts E½Ai
are given by Eqs. (8) and (9), respectively [15]. It is worth mentioning that CW j represents the size of the contention window
while pi represents the collision probability.
X
m1 X
k
CW j X
m
CW j
E½W i ¼ pki ð1 pi Þ þ pm
i ð8Þ
k¼0 j¼0
2 j¼0
2
X
m1
E½Ai ¼ pki ð1 pi Þðk þ 1Þ þ pm
i ðm þ 1Þ ð9Þ
k¼0
Based on the expected wait off time and expected number of transmission attempts, transmission probability T i is given
by Eq. (10).
E½Ai
Ti ¼ ð10Þ
E½W i þ E½Ai
IEEE 802.11n transmits frame on the physical layer level which can be modelled by incorporating SIFS, DIFS, frame trans-
mission time and acknowledgement time. Let T s be the time taken by a frame if it is transmitted successfully. Let T c be the
time if a frame is not transmitted successfully. T s and T c can be modelled by viewing the frame transmission time
T frame ; SIFS; T ack and DIFS as given in Eqs. (11) and (12).
T s ¼ T frame þ SIFS þ T ack þ DIFS ð11Þ

T c ¼ T frame þ T acktimeout ð12Þ
Also, collision time is a function of collision probability to success probability. Based on [15], average collision time T av g ðcÞ
can be derived from the packet loss probability pi and packet collision time T c as shown in Eq. (13).
pi
T av g ðcÞ ¼ Tc ð13Þ
1 pi
Eqs. (3)–(13) are used to evaluate the throughput using collision time and transmission probability. Analytical results for
capacity of IPTV and VoIP using UDP are shown in Table 3. Results show that IPTV and VoIP users have an inverse relation-
ship with each other. This behaviour shows that IPTV users occupy majority share in bandwidth which reduces share for
VoIP users. Network congestion also increases with large number of IPTV users which increases delay for further VoIP users.
Similar trends are observed experimentally for IPTV and VoIP users. Results from Table 3 show that 4 IPTV users can be
accommodated with 1 VoIP user maximally. All of our results represent the steady state conditions where throughput of
all supportable IPTV and VoIP streams is at saturation level. It is important to mention that all of our simulation, experimen-
tal and analytical results differ by a small amount based upon the following reasons.
Analytical results show maximum capacity because all analytical results present a mean estimate of capacity of IPTV and
VoIP users based upon their bandwidth requirements in wireless medium.
Simulations present an estimate of users obtained by simulating the environment in ns2. Collisions at a particular instant
decrease the capacity in simulations.
Experimental results show minimum capacity because of the particular environment, e.g. room walls, obstacles, transmit-
ting device location and receiving device location and reception system, etc. deteriorate the capacity from the ideal ana-
lytical capacity.
IPTV using UDP encounters packet loss at the AP because UDP is a constant bit rate protocol and packets drop from queues
of AP. UDP cannot cope fairly with the network congestion and maintains its data rate at any situation. To resolve congestion
less mechanism of UDP, DCCP has been proposed. DCCP has two variants namely TCP-like and TFRC.
TCP-like uses the congestion control mechanism similar to TCP. TCP-like has no retransmissions. TCP-like applies conges-
tion control to acknowledgements, it works on units of packets instead of bytes and TCP-like uses no retransmissions. These
Table 3
IPTV and VoIP capacity over IEEE 802.11n using UDP.
Simulation Analytical Experimental

IPTV VoIP IPTV VoIP IPTV VoIP
1 36 1 39 1 32
2 24 2 28 2 21
3 11 3 14 3 8
4 2 4 6 4 1
features make it different from TCP but in reliability it resembles TCP because it decreases its data rate much more sharply in
congested situations similar to TCP.
We have shown previously that TCP-like gives poor performance for IPTV and VoIP because TCP-like decreases its data
rate in congested situations. Decrease in data rate increases the delay for all packets reaching the receiver end. IPTV is a real
time service with limitations of packet loss and delay. Large delay disrupts the support for many users.
On contrary to TCP-like, TFRC is the protocol which resembles UDP in some aspects like no retransmissions and less reli-
ability. Instead of constant bit rate, TFRC adjusts its data rate continuously according to the packet loss rate and RTT expe-
rienced at the user end. Any increase in packet loss or RTT implies network congestion which results in throughput reduction
of TFRC. TFRC increases its data rate in low packet loss and RTT conditions.
Let s; p; R and t RTO be the segment size, packet loss rate, round trip time and TCP retransmission timeout value, respec-
tively. Let b be the packets acknowledged by a single TCP acknowledgement. Using above parameters, throughput of TFRC is
modelled by the Eq. (14). It is pertinent to mention that we define throughput as the physical layer data rate sent by the
transmitter on the physical link. Throughout this paper, we add the various standard header fields into the transport layer
header to estimate the physical layer throughput.
s
X ¼ qffiffiffiffiffiffi qffiffiffiffiffiffi ð14Þ
R 3 þ tRTO 3 3bp
2bp
8
pð1 þ 32p2 Þ
Analytical results of throughput for a range of packet loss probabilities and RTT are shown in Fig. 2. RTT trends show that
RTT is inversely related to the data rate. Packet loss trends show that packet loss is inversely related to the throughput gov-
erned by TFRC. Moreover, trends of TFRC are more drastic for change in packet loss than change in RTT.
Table 4 shows the capacity of IPTV and VoIP over IEEE 802.11n using TFRC for both applications. Results show that TFRC
can accommodate more IPTV users than UDP. TFRC adjusts its data rate by decreasing its data rate in congested situations.
Coping with the network situations makes TFRC a better suitable candidate for transmission of IPTV and VoIP with increased
capacity. Analytical and experimental results show that IPTV and VoIP with TFRC can accommodate at least 5 IPTV users with
0 VoIP users. This suggests that TFRC provides 1 more HDTV user than UDP. Comparison of Tables 3 and 4 shows that UDP
and TFRC provide same VoIP capacity for small number of IPTV users. On contrary, large number of IPTV users can be accom-
modated with the use of TFRC. This suggests that TFRC is more suitable for IPTV than VoIP. VoIP has small packet size with
different packetization interval than IPTV. This suggests that VoIP suits well to UDP than TFRC. In next subsection, we aim to
figure out IPTV and VoIP running different protocols either UDP or TFRC.
4.2. Cross-protocol performance of IPTV and VoIP using UDP and TFRC
In this section, we analyze the performance of IPTV and VoIP over the protocols UDP and TFRC. Based on the current
network architecture, both IPTV and VoIP use UDP at the transport layer. Aim of this section is to investigate: How about
changing only IPTV or VoIP transport layer protocol without influencing other network traffic?
We evaluate the performance of TFRC based IPTV in presence of UDP based VoIP. UDP is a constant bit rate protocol.
Available capacity in network decreases with UDP because TFRC decreases its data rate based on the increase in packet loss
rate and RTT. On contrary, UDP keeps sending packets with constant bit rate. Our analysis shows that data rate of TFRC is
highly dependent upon the bit rate of UDP. To increase the TFRC based IPTV connections, UDP connections must be reduced.
A comparison of analytical and experimental performance versus simulation results is shown in Table 5.
7 7
x 10 x 10
p=0.000021
7.5 RTT=41ms
p=0.000023 8
RTT=45ms
p=0.000019
Throughput (bps)
Throughput (bps)
7 7 RTT=49ms
6.5 6
6 5
4
5.5
3
5
2
0.04 0.045 0.05 0.055 0.5 1 1.5 2
Round Trip Time (RTT) (ms) Packet Loss Probability (p)
x 10−4
Fig. 2. TFRC round trip time and packet loss probability versus throughput.
Table 4
IPTV and VoIP capacity over IEEE 802.11n using TFRC.

1 35 1 39 1 32
2 22 2 26 2 20
3 11 3 17 3 8
4 2 4 6 4 1
5 1 5 4 5 0
Table 5
IPTV(TFRC) and VoIP(UDP) capacity over IEEE 802.11n using TFRC.

1 7 1 9 1 5
2 5 2 6 2 4
3 3 3 4 3 1
4 0 4 2 4 0
Similarly, we evaluate the performance of UDP based IPTV with TFRC based VoIP. Table 6 shows the analytical and exper-
imental results versus simulation results. Results show that TFRC follows same trends in presence of UDP. Number of UDP
connections must be reduced in order to avoid network congestion. Results show UDP and TFRC cannot co-exist fairly with
each other because UDP occupies all the network bandwidth irrespective of the packet loss and delay encountered by the
system.
4.3. Fairness analysis of IPTV and VoIP with TCP traffic
Studies show that 80% of network traffic is non-real time traffic [12]. This suggests that fairness of IPTV and VoIP traffic
must be studied with Transmission Control Protocol (TCP) traffic in the network. To implement TCP traffic, we use a packet
size of 1000 bytes with File Transfer Protocol (FTP) at application layer. TCP is used at the transport layer of FTP which makes
FTP a suitable candidate to test non-real time reliable traffic. Multiple TCP connections are established because probability of
achieving minimum window size is highest with a single TCP source.
Capacity and fairness results from analytical and experimental evaluations of IPTV and VoIP traffic using UDP with FTP
traffic are shown in Table 7. Results show that TCP decreases its window size in presence of constant bit rate protocol
Table 6
IPTV(UDP) and VoIP(TFRC) capacity over IEEE 802.11n.

1 8 1 10 1 6
2 6 2 7 2 4
3 4 3 5 3 3
4 1 4 2 4 0
Table 7
Performance statistics for combined IPTV and VoIP along with TCP traffic.
IPTV Combined flows Average throughput (Mbps)

HDTV UDP- UDP: 50.7 UDP: 51 UDP: 47.2
TCP TCP: 6.8 TCP: 7 TCP: 4.3
HDTV TFRC- TFRC: 66.5 TFRC: 68.2 TFRC:62.3
TCP TCP: 14.2 TCP: 16.3 TCP: 11.5
SDTV UDP- UDP: 51.3 UDP: 52.9 UDP: 48.4
TCP TCP: 6.1 TCP: 8.4 TCP: 4.2
SDTV TFRC- TFRC:64.7 TFRC: 67.3 TFRC:61.2
TCP TCP: 14.4 TCP: 16.2 TCP: 12.3
UDP. This suggests that UDP provides not only less capacity but also provides unfair share in bandwidth to TCP traffic in the
network.
We test the performance of IPTV and VoIP using TFRC with FTP traffic in the network as shown in Table 7. Results show
that TCP competes fairly with TFRC traffic in the network. Bandwidth analysis shows that TFRC provides more share in band-
width to TCP traffic than UDP. TCP gets 4.3 Mbps throughput in presence of UDP while 11.5 Mbps throughput in presence of
TFRC experimentally. Comparable performance gains with similar trends are observed for both SDTV and HDTV. TFRC pro-
vides at least 167.4% more throughput to TCP than UDP.
Our study over various transport layer protocols suggests that TFRC provides better capacity than UDP for transmission of
IPTV and VoIP over IEEE 802.11n. TFRC adjusts its data rate according to the network conditions which makes it a better can-
didate for IPTV and VoIP. Our investigation suggests that TFRC must be adopted for all real time applications. Performance of
TFRC deteriorates severely in presence of UDP due to congestion less mechanism of UDP. TFRC provides much more fair share
in bandwidth to TCP than UDP. In the next section, we aim to investigate the optimum physical layer parameters for com-
bined IPTV and VoIP over IEEE 802.11n.
5. Optimal physical layer parameters of IEEE 802.11n for IPTV and VoIP
IEEE 802.11n is equipped with a number of parameters and enhanced features including queue size, SIFS, DIFS, contention
window size and physical layer header time. Simulation results of our previous investigations [3,4] for transmission of IPTV
and VoIP are shown in Table 8. In this section we present the analytical and experimental evaluation of all the parameters.
5.1. Trends of SIFS, DIFS and physical header
Layers of computer network protocols are independent of each other. So, all layers can be modelled independently.
Parameters like SIFS, DIFS, contention window and physical header time belong to the physical layer. It means that any
change in physical layer parameters needs to be modelled at only physical layer level.
Let T SIFS be the time duration required to wait for SIFS time, T DIFS be the time duration required to wait for DIFS time, T PHY
be the time duration required for physical header transmission and T payload be the time required to transmit payload data.
Various time periods as observed at the physical layer are shown in Table 1. Throughput at physical layer given by X can
be modelled by Eq. (15) assuming RTS/CTS is disabled. Eq. (16) shows the throughput equation if RTS/CTS is enabled.
s
X¼ ð15Þ
T SIFS þ T DIFS þ T payload þ T header þ T ack
s
X¼ ð16Þ
3T SIFS þ T DIFS þ T payload þ T header þ T ack
Using default values of all time periods, various parameters are varied in Eqs. (15) and (16). Fig. 3 shows the results of
simulations and analytical observations by varying various parameters. Results show that SIFS and DIFS are inversely pro-
portional to the throughput. Moreover, SIFS has more drastic effect on throughput than DIFS because SIFS is encountered
more than DIFS. SIFS and DIFS have no optimal values but DIFS must have the value given by Eq. (17). T Proptime is the prop-
agation of a packet from AP to user.
T DIFS ¼ 2T SIFS þ T Proptime ð17Þ

If DIFS is less than SIFS then throughput would be zero effectively because all stations would transmit when any station is
waiting for SIFS. This results in wastage of capacity in form of collisions. Trends of physical header time with RTS/CTS
enabled and disabled are shown in Fig. 4. Decrease in physical header time provides more throughput because remaining
time is used for payload transmission. We suggest a decrease of only 10% which changes physical header time to 18 ls. Large
change in physical header is not possible due to small clocking frequency of devices.
Results show that decrease in time durations of physical layer parameters gives more time for data transmission. We sug-
gest a decrease of only 10% in physical header duration which can increase capacity by at least 1 VoIP user. Our previous
simulations [4] also proposed 10% decrease in parameters through simulations only. Large decrease in physical layer param-
eters is not possible due to limitation of physical devices.
Table 8
Parameters for IEEE 802.11n.
Parameters Default parameters Proposed parameters [4]

Queue size 50 pkts 70 pkts
SIFS 16 ls 14.4 ls
DIFS 34 ls 30.6 ls
Physical header 20 ls 18 ls
Contention window 15 11
8 SIFS behaviour 7 DIFS behaviour

x 10 x 10
9
2 RTS/CTS enabled RTS/CTS disabled
RTS/CTS disabled 8 RTS/CTS enabled
Throughput (bps)
Throughput (bps)
1.5
7
6
1
5
0.5
4
0 1 2 3 2 3 4 5
−5 −5
SIFS (µs) x 10 DIFS (µs) x 10
Fig. 3. Trends SIFS and DIFS with RTS/CTS enabled and disabled.
7
x 10
11
RTS/CTS enables
RTS/CTS disabled
10
Throughput (bps)
6
1 1.5 2 2.5 3 3.5
−5
Physical header time (µs) x 10
Fig. 4. Trends physical header time with RTS/CTS enabled and disabled.
5.2. Trends of contention window
Contention window represents the physical layer waiting time encountered in CSMA/CA which tries to avoid collisions
and also tries its best to minimize the redundant waiting time. Our previous study [4] has shown through simulations that
contention window size of 11 is the optimal size instead of 15. Large packet size is the major cause for small contention win-
dow size. IPTV has got a large data rate requirement but the packet size is 1366 bytes. Large data rate limits the support for
large number of IPTV users. This suggests that only few IPTV users are competing with each other at any particular instant.
Few competing users can accommodate in a situation of small window size as compared to large window size. Large con-
tention window size results in redundant waiting times of all users which decreases throughput and capacity.
Considering the CSMA/CA backoff mechanism, various transmission probabilities have been shown in [15]. Analytical
results for IPTV and VoIP from CSMA/CA equations are shown in Fig. 5. Trends show that throughput increases upto a con-
tention window size of 11. This behaviour suggests that small contention window size results in collision of packets between
various users. Beyond contention window size of 11, throughput decreases. This suggests that large contention window size
results in redundant waiting time of stations which decreases throughput slightly. Our analytical and simulation results con-
firm that contention window size of 11 is the optimal size. Results suggest that only optimal contention window size should
be used for practical applications running IPTV and VoIP. Moreover, for large scenarios, contention window size can be esti-
mated practically by computing the throughput obtained by all devices at varying contention window sizes. Contention win-
dow size providing maximum throughput is the optimal size for that particular configuration of various devices. It is
pertinent to mention that slight tuning of contention window is required for applications running TCP traffic simultaneously
with IPTV and VoIP traffic based upon the number of competing devices.
70
60
Throughput (Mbps)
50
40
30
20
10 Analytical
Simulation
0
0 5 10 15
Contention Window (CW)
Fig. 5. Contention window trends for IPTV and VoIP.
5.3. Trends of queue size
Queue size represents the physical layer buffering capacity for the wireless AP. Queue size for IPTV and VoIP over IEEE
802.11n can be modelled through queue utilization ratio. Let traffic arrival rate be Ai and departure rate be Di , then queue
utilization ratio q is given by Eq. (18).
Arriv alRate Ai
q¼ ¼ ð18Þ
DepartureRate Di
Actual queue size for any system is a function of the queue utilization ratio. Queue utilization ratio of less than 1 indicates
small arrival rate as compared to serving rate. On contrary, queue utilization ratio greater than 1 indicates less serving rate
than the utilization rate. Queue size, given by B, can be represented as a function of queue utilization ratio q by Eq. (19).
1q B
PðQueueSize ¼ BÞ ¼ q ð19Þ
1 qB1
Eq. (19) can be simplified arithmetically into Eq. (20).
1
qB ¼ ð20Þ
1 q þ Pq
Queue size depends upon the traffic arrival rate. This suggests that it needs to be modelled with different traffic catego-
ries. Traffic arrival rate of TCP and TCP-like resembles Additive Increase and Multiplicative Decrease (AIMD) model. A sim-
plified model for estimating TCP rate is given by Eq. (21).
rffiffiffiffiffiffi
1 3
R¼ ð21Þ
RTT 2P
Traffic arrival rate of TFRC is modelled by the TFRC rate equation. Eq. (22) shows the rate equation of TFRC.
s
X ¼ qffiffiffiffiffiffi qffiffiffiffiffiffi ð22Þ
R 3 þ tRTO 3 3bp
2bp
8
pð1 þ 32p2 Þ
Our analysis shows that the bottleneck link in IPTV transmission is the wireless AP. TFRC increases its data rate until it
gets some packet loss or increased RTT from the AP. On contrary to TFRC, UDP is a congestion-less protocol which provides
constant bit rate. Fixed data rate of UDP provides a fixed arrival rate at the queues of AP and queue size depends upon the
transmission rate from queues of AP to the wireless user.
To model the queue size of UDP, we use a constant bit rate model having uniform distribution. IEEE 802.11n provides a
theoretical data rate of 600 Mbps while a practical data rate of 300 Mbps at physical layer level using 3 3 MIMO
technology.
Fig. 6 presents the analytical and experimental results for optimal queue size of IPTV and VoIP for transmission over IEEE
802.11n. Results show that initially queue size increases throughput sharply. After 100 packets, increase in throughput is
nearly negligible. Results show that maximum capacity is obtained at a queue size of 70 packets. Very large queue size
results only in wastage of resources because wireless channel becomes the bottleneck link for large queue size. Experimental
results show less throughput than analytical because of varying network conditions and large packet loss in a real network
scenario. Delay is maximum for experimental results because of the wireless conditions. Analytical results display minimum
delay because they incorporate limited network conditions. Simulation results display delay and throughput in between
experimental and analytical.
5.4. Trends of aggregation
IEEE 802.11n is equipped with the aggregation mechanisms which use aggregation at two levels. At level 1, multiple MAC
layer frames called MAC Service Data Units (MSDUs) are aggregated together to form an Aggregated MAC Service Data Unit
(A-MSDU). At level 2, multiple physical layer frames composed of physical layer header and A-MSDUs are aggregated
together to transmit an Aggregated MAC Physical Data Unit (A-MPDU).
Let T MPDU be the transmission time of a single MPDU and N be the number of MPDUs inside a single A-MPDU. Let T SIFS and
T DIFS be the time required to wait for SIFS and DIFS before transmission. Then transmission time T of an MPDU (without
aggregation) is given by Eq. (23). Eq. (24) represents the transmission time T AMPDU with an aggregation of N MPDUs inside
a single A-MPDU.
T ¼ T DIFS þ T MPDU þ T SIFS þ T ACK ð23Þ

T AMPDU ¼ T DIFS þ NT MPDU þ T SIFS þ T ACK ð24Þ
Fig. 7 shows the aggregation performance of IPTV and VoIP over IEEE 802.11n. Results show that capacity increases upto 4
times aggregation due to reduction in collision time and header overhead. Beyond 4-times aggregation, large delay disrupts
the support for large number of users.
70 52
50
60
48
Throughput (Mbps)
50
Delay (msec)
46
40
44
30
42
Analytical Analytical
20
Simulation 40 Simulation
Experimental Experimental
10 38
0 200 400 0 200 400
Queue Size (Pkts) Queue Size (Pkts)
Fig. 6. Queue size trends for IPTV and VoIP.
Fig. 7. Aggregation trends for IPTV and VoIP.
Comparison of physical layer parameters for IPTV and VoIP over IEEE 802.11n shows that maximum capacity can be
achieved at optimal values of all the parameters. Table 9 presents the comparison of optimal parameters of IPTV and VoIP
over IEEE 802.11n. As layers are independent so changes in physical layer parameters do not affect transport layer protocol.
Optimal parameters remain same with the use of UDP or TFRC. SIFS, DIFS and physical header have no optimal values
because decrease in physical layer parameters gives more time for payload data to transmit. This shows that SIFS, DIFS
and physical header duration must be reduced as much as possible. We suggest a decrease of 10% in these parameters which
can increase capacity by at least 1 VoIP user without effecting current network.
6. Improving performance through multicast mechanism
In this section, we aim to analyse and improve the performance of combined IPTV and VoIP over IEEE 802.11n. In this
regard, we present the limitations of current IPTV architecture using TFRC. We analyse the performance of IPTV in presence
of VoIP using multicast TFMCC protocol. Finally, we present our proposed protocol W-TFMCC by mitigating the limitations of
all previous protocols. We present the performance of W-TFMCC with a test run real network scenario and explain its per-
formance gains.
6.1. Using multicast for IPTV transmission
Studies reveal that capacity of IPTV is dependent upon the channel popularity [25]. Zipf’s law states that the channel
viewership of xth channel would be x-times less than the first channel [25]. Fig. 8 shows the histograms of channel viewer-
ship versus channel popularity.
Trends of histogram show that channel viewership decreases exponentially as its popularity decreases. So, unicast mech-
anism for transmission of IPTV results in wastage of capacity. Our results for IPTV capacity using UDP and TFRC show that
multiple receptions of same channel require multiple transmissions. This suggests that multicast transmissions must be used
for all users viewing same channel. TFRC and TCP-like employ unicast mechanism to observe the packet loss and RTT of the
receiver. Our study shows that congestion control mechanism of TFRC can enhance capacity of IPTV. For multicast mecha-
nisms, TCP Friendly Multicast Congestion Control Protocol (TFMCC) has been designed which has congestion control mech-
anisms similar to TFRC. TFMCC selects the worst case receiver based on the packet loss rate and makes all transmissions
based on the worst case receiver. It was believed in TFMCC that all receivers would get the reception if the worst case recei-
ver gets the reception. Our analysis shows that performance of TFMCC is limited in wireless medium. In wireless medium,
TFMCC selects the worst case receiver and adjusts its data rate according to the packet loss rate and RTT experienced by the
worst case receiver. This suggests that any user experiencing worst-case receiver conditions deteriorates the performance of
all other users viewing same channel. Fig. 9 presents the scenario in which few IPTV users lie inside first range of AP having
small packet losses while other users lying outside first range experiencing large packet loss conditions.
Table 9
Optimal parameters of IEEE 802.11n.
Parameters Standard values Simulations [4] Analytical Experimental

Queue Size 50 pkts 70 pkts 75 pkts 80 pkts
Aggregation 1 pkt 4 pkts 4 pkts 4 pkts
Contention window 15 11 11 11
0.25
0.2
Viewership
0.15
0.1
0.05
0
2 4 6 8 10 12 14 16 18 20
Channels Popularity
Fig. 8. Channel viewership with popularity (Zipf’s Law).
Fig. 9. Network scenario for IPTV and VoIP.
In this paper, we propose wireless enhancement of TCP Friendly Multicast Congestion Control Protocol (W-TFMCC). Basic
idea of W-TFMCC is to limit the capacity of only those users experiencing worst case conditions. Throughput of all users lying
in better environment must not be deteriorated. W-TFMCC estimates the packet loss rate of all users lying in its range and
makes an estimate of all users which can be provided with a reliable QoS either HDTV streams or SDTV streams. We show
that capacity of W-TFMCC provides 61% coverage area in comparison to TFMCC and TFRC providing only 30% and 5% cover-
age areas respectively. Moreover, W-TFMCC results not only in an increase in capacity but also saves resources by avoiding
any redundant transmissions of channels which makes it more efficient.
6.2. TFMCC for IPTV transmission
TFMCC is a single rate multicast congestion control protocol. It competes fairly with TCP traffic in the network because it
uses congestion control mechanism similar to TFRC. TFMCC uses the packet loss rate of the receivers to determine the send-
ing rate continuously. TFMCC protocol uses an equation to determine the sending rate continuously. Let p be the packet loss
rate and s be the segment size, then TFMCC throughput X is given by Eq. (25).
8s
X ¼ qffiffiffiffi qffiffiffiffi ð25Þ
R 3 þ 12 3p
2p
8
pð1 þ 32p2 Þ
Simulations for TFMCC are performed over ns2 by extending support of [4] for IPTV and VoIP over IEEE 802.11n.
Performance of TFMCC has been tested in varying environments as shown in Table 10. Different regions have different range
of packet loss and RTT. Our results for performance analysis of TFMCC over IEEE 802.11n for transmission of IPTV and VoIP
are shown in Table 11. Coverage area shows the percentage of supported users according to Zipf’s law. Results show that
TFMCC gives a high coverage area 61.65% in Terrain-A with ideal conditions. However, coverage area reduces to 30% in worst
conditions which suggests that only 30 users are supported out of 100 (approximately for a certain bandwidth). Performance
of TFMCC is worst in varying network conditions. TFMCC gives increased performance over IEEE 802.11n in reliable network
conditions where all users have limited packet loss and RTT. Performance of TFMCC deteriorates in large packet loss and RTT
conditions. Different users lying in different conditions demanding same channel get worst QoS because TFMCC selects worst
case receiver. This necessitates the demand of W-TFMCC to provide better QoS to all users.
6.3. Wireless enhancement of TFMCC (W-TFMCC)
Our proposed protocol W-TFMCC is an enhancement of current protocol TFMCC. Basic limitation in current mechanism
arises from the varying channel conditions in wireless environment.
Table 10
Terrains for IPTV and VoIP over IEEE 802.11n.
Terrain Environment
Type-A HDTV Support : Low Packet Loss and Low RTT
Type-B HDTV Support : Low Packet Loss and Large RTT
Type-C SDTV Support : Large Packet Loss and Low RTT
Type-D SDTV Support : Large Packet Loss and Large RTT
Type-E SDTV Support : Very Large Packet Loss and RTT
Table 11
IPTV and VoIP capacity over IEEE 802.11n using TFRC and TFMCC respectively.
Terrain TFRC TFMCC Coverage %

Type Simulation Simulation Analytical Experimental
TV VoIP TV VoIP TV VoIP TV VoIP
A 5 1 1 35 1 39 1 32 61.65
B 4 15 1 33 1 37 1 29 54
C 4 8 1 25 1 29 1 23 49
D 4 3 1 14 1 19 1 12 44
E 3 5 1 6 1 9 1 4 30
TFMCC transmits all streams using a single multicast transmission depending upon the worst case user reception.
Transmissions to worst case user deteriorate the data rate and effectively throughput is worstly effected for all users. On
contrary, W-TFMCC selects the worst case user depending upon the packet loss conditions and RTT experienced by worst
case user. If HDTV throughput is not possible for any user then W-TFMCC transmits a separate channel group for users hav-
ing large packet loss.
Based upon our previous investigations on TFRC, we prefer the same data rate equation of TFRC to be used for W-TFMCC,
as shown in Eq. (26).
8s
X ¼ qffiffiffiffi qffiffiffiffi ð26Þ
R 3 þ 12 3p
2p
8
pð1 þ 32p2 Þ
If packet loss or RTT or both parameters increase beyond a limit then W-TFMCC separates those users from current mul-
ticast transmission. Fig. 10 presents the scenarios of wireless environment with varying loss and RTT conditions. Limits of
packet loss and RTT for provision of HDTV and SDTV streams with reliable QoS are marked in the figure.
Our simulation and analytical results for W-TFMCC performance in comparison to TFRC are shown in Table 12. Results
show that performance of TFRC deteriorates in worst conditions because supported users are limited. On contrary, W-TFMCC
transmits streams to group of users. For ideal channel conditions, like Terrain-A, W-TFMCC transmits only 1 stream instead
of 5 for 5 users watching same channel. To estimate the performance of W-TFMCC, we use the users distribution as given by
Zipf’s law. Considering a set of 20 channels, probability of watching first five channels is the sum of individual channels
0.25
HDTV
SDTV
0.2
Round Trip Time (s)
0.15
0.1
0.05
0
0 0.002 0.004 0.006 0.008 0.01
Packet Loss Probability
Fig. 10. Packet loss and RTT limits for SDTV and HDTV channels.
Table 12
IPTV and VoIP capacity over IEEE 802.11n using TFRC and W-TFMCC respectively.
Terrain type TFRC TFMCC Coverage %

Simulation Simulation Analytical Experimental
TV VoIP TV VoIP TV VoIP TV VoIP
A 5 1 1 35 1 39 1 32 61.65
B 4 15 2 30 2 34 2 26 61.65
C 4 8 2 22 2 26 2 20 61.65
D 4 3 2 12 2 18 2 15 58.7
E 3 5 2 3 2 7 2 6 56.25
probabilities. To support 5 users watching same channel, TFRC transmits 5 streams. On contrary W-TFMCC transmits only 1
stream for 5 users watching same channel. Using Zipf’s law, coverage area of W-TFMCC appear to be 61.65%. This suggests
that 61 IPTV users can be supported out of 100 users which is a remarkable gain. In worst terrains, coverage area decreases to
56.25% users because SDTV and HDTV streams have to be transmitted simultaneously to support different group of users
watching same channel. Results show that W-TFMCC transforms the concept of capacity from number of users to number
of channel streams. In case of excess packet loss and RTT, W-TFMCC stops transmissions until it finds the ideal situation
for SDTV or HDTV transmissions. Capacity of W-TFMCC is always greater or equal to TFRC and TFMCC. Worst case of W-
TFMCC can occur if all users are present in different conditions and viewing different channels. In such a situation, W-TFMCC
would converge to unicast mechanism. This suggests that W-TFMCC can perform better in all environments. Fig. 11 presents
the coverage comparison for 100 users using UDP/TFRC/TFMCC/W-TFMCC. Results show that W-TFMCC performs best in
worst terrains. Large number of users can be supported easily using W-TFMCC. Advantage gain increases with more users
in comparison to TFRC.
Algorithm 1 presents the pseudocode for performance of W-TFMCC. X represents the throughput which is estimated
using the RTT and packet loss probability. X hd and X sd denote the reference throughputs required to transmit HDTV and SDTV
streams respectively as given by Table 2. Repeated measurements are performed after every finite amount of time. Sampling
time must be as small as possible to detect the continuously varying wireless conditions. Users groups are formed based
upon RTT and packet loss probability.
Algorithm 1. W-TFMCC protocol.
1 pusr ðiÞ; /⁄ Packet loss probability of user i ⁄/

2 RTTðiÞ; /⁄ RTT of user i ⁄/
3 HDch =[]; /⁄ Transmitted HDTV channels list ⁄/
4 SDch =[]; /⁄ Transmitted SDTV channels list ⁄/
5 HDðiÞ; /⁄ HDTV channel for user i ⁄/
6 SDðiÞ; /⁄ SDTV channel for user i ⁄/
s ffiffiffiffiffi
7 XðiÞ ¼ pffiffiffiffiffi
2bp
p
3bp
;
R 3
þt RTO 3 8
pð1þ32p2 Þ
8 if XðiÞ > X hd then
9 HDch = [HDch HDðiÞ]; /⁄ Start HDTV transmission for i ⁄/
10 else if XðiÞ > X sd then
11 SDch = [SDch SDðiÞ]; /⁄ Start SD transmission for i ⁄/
12 else
13 HDch = HDch - pusr ðiÞ; /⁄ Remove user from channel list ⁄/
14 SDch = SDch - pusr ðiÞ;
15 end
6.4. Test run for W-TFMCC
Performance of W-TFMCC can be analysed with a test run having challenging wireless conditions. It is pertinent to men-
tion that Fig. 12 presents a real home network scenario. Wireless AP is located in one room depending upon the available
sockets and users convenience. There are four regions depending upon the signal strength in various regions. Various regions
have been marked with different patterns. We analyse the performance of all protocols in the given test run environment.
Fig. 11. Coverage comparison of UDP/TFRC/TFMCC/W-TFMCC.
Fig. 12. Test run scenario for IPTV and VoIP using W-TFMCC.
Depending upon the packet loss and RTT, our experimental receiver device splits coverage into four different regions from
Level-1 to Level-4. We consider users demand in each region such that Level-1 region demands an HDTV and a FTP device
(computer). There are three regions (rooms) having Level-2 coverage. Level-2 demands two HDTV, three VoIP phones and an
FTP device (laptop). Level-3 region demands two HDTV and one VoIP phone. Level-4 coverage area has a single HDTV
demand.
UDP and TFRC use unicast transmission mechanisms. Demand of each user is fulfilled by individual and unique transmis-
sion with UDP/TFRC. Our investigation reveals that 4 IPTV users can work using UDP while 5 HDTV devices can work with
TFRC. Unicast transmissions of UDP/TFRC are independent of each other. All users are supplied with individual streams even
if all users are viewing same channel. This suggests that UDP and TFRC waste significant resources because of unicast
mechanism.
TFMCC uses the multicast mechanism by supplying single channel stream to all users viewing same channel. Major lim-
itation of TFMCC arises when user in level-4 and level-1 are viewing same channel. TFMCC selects the level-4 user because it
is in worst condition as compared to user in level-1 region. All transmissions are performed according to level-4 user. As a
result, users lying in level-1 suffer poor image quality because TFMCC adjusts its data rate according to level-4 user. Worst
conditions of level-4 user deteriorate the performance of level-1 user who is unable to watch HDTV stream.
Our proposed protocol W-TFMCC gives best performance in this environment. Packet loss and RTT of users lying in level-1
and level-4 regions watching same channel is identified. If packet loss and RTT of level-4 user is greater than the possible
limit for HDTV stream then level-4 user is supplied with an SDTV stream. Enhanced group-based mechanism has sorted
the problem of level-1 user by supplying it with an HDTV stream. Channels of all users lying in different regions are iden-
tified with W-TFMCC. Packet loss and RTT of all users watching same channel are identified. Various channel streams are
supplied to different groups based upon their environment conditions. Worst case of W-TFMCC is the scenario in which
all users are lying in different conditions and watching different channels. In such a case, performance of W-TFMCC would
be similar to TFRC because it would use unicast streams for all viewers.
6.5. Comparison of TFMCC vs W-TFMCC
Comparison of TFMCC with W-TFMCC shows that our proposed W-TFMCC protocol enhances capacity in tough terrains.
Among the five tested terrain types (A, B, C, D, E), which varied in packet loss and round trip time conditions, significant per-
formance gains are achieved for W-TFMCC. Table 11 shows that coverage of TFMCC drops from 61.65% to 30% by moving
from terrain A to E. On contrary, our proposed protocol W-TFMCC drops coverage from 61.65% to only 56.25% by moving
from terrain A to E, as shown in Table 12. Moreover, W-TFMCC provides 61.65% coverage in slight tough terrains (B, C) too.
Major reason for performance enhancement of W-TFMCC is attributed to group based channel distribution.
7. Prior state-of-the-art approaches
In this section, we present the performance results of previous state of the art approaches. In [2], authors enhance the
quality of experience (QoE) by investigating the most appropriate technology for IPTV. Authors conclude that IEEE
802.11a provides the best performance preceded by WIMAX and IEEE 802.11g network. In [6], authors show that 6–7 video
sources over 1-hop and 2–3 video sources over 3-hop wireless network are supportable. In [7], authors show through exper-
iments that IPTV over IEEE 802:11n can support 12 and 10 users in indoor and outdoor environments, respectively. In [5],
authors show that IEEE 802:11b and IEEE 802:11g networks can support 2 and 6 streams while IEEE 802:11n can support
dozens of IPTV streams. In [8], authors perform IPTV experiments and show the commercial aspects of wireless IPTV. In
[14], authors show through experiments that three SDTV channels with at least one VoIP stream can be supplied over
two hops in IEEE 802.11 b/g access points. In [20], authors show through ns2 simulations that multimedia applications
can be provided with less delay, less jitter and less packet loss with their proposed protocol. In [9], authors verify through
simulations that a protocol in between TCP and UDP at transport layer can provide better streaming performance. Compar-
ison of various studies shows that performance of previous studies was limited owing to less throughput of predecessor IEEE
802.11 standards. Although TFRC has provided better performance but it becomes less useful for IPTV which requires mul-
ticast wireless transmissions. Our proposed protocol W-TFMCC has shown to support 32 VoIP users with 1 multicast IPTV
user (experimentally) which is an unprecedented effort. Major novel improvements in our analysis include the analytical
and experimental investigations of optimal transport layer protocols with optimal physical layer parameters for transmis-
sion of combined IPTV and VoIP over IEEE 802.11n, which has not been covered in previous literature.
8. Conclusion
In this paper, we evaluate the capacity of IPTV and VoIP over IEEE 802.11n experimentally and analytically. Our investi-
gation over transport layer protocols shows that use of TFRC instead of UDP at transport layer can enhance IPTV capacity by
25%. TFRC adjusts its data rate according to the congestion situations in the network which makes it highly suitable for IPTV.
Results show that TFRC provides fair share in bandwidth to TCP traffic than UDP. Our investigation over physical layer
parameters of IEEE 802.11n shows that queue size and contention window have optimal size of 70 pkts and 11 respectively
for IPTV and VoIP users. Study shows that optimal values of SIFS, DIFS, Physical header can increase capacity by 10% at least.
Trends of aggregation show an optimal aggregation of 4 packets for IPTV and VoIP beyond which large delay disrupts the
support for many users. Our major contribution is the proposition of W-TFMCC protocol with extensive simulations and
experiments. Performance of TFMCC protocol deteriorates in severe wireless conditions due to absence of group based trans-
mission mechanism. W-TFMCC protocol provides multicast HDTV and SDTV streams to users based upon the delay and RTT
of packets of all users. W-TFMCC incorporates not only number of users but deals with the channel viewership too. Number
of streams transmitted by W-TFMCC are directly related to the number of channels watched by users. Results show that per-
formance of W-TFMCC is greater than TFRC/TFMCC if at least two users are watching the same channel. W-TFMCC adjusts its
data rate according to the network conditions by making user groups depending upon the packet loss conditions. Our study
concludes that use of W-TFMCC with optimal physical layer parameters can increase network capacity at least by 44% in
comparison to UDP and TFRC respectively.
References
[1] Cisco. Cisco visual networking index forecast 2011–2016 <http://goo.gl/hSJjX>.

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Vegas (USA); September 2013.
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Saad Saleh received his MS and BS degrees in Electrical Engineering from National University of Sciences and Technology (NUST), Islamabad in 2013 and
2011, respectively. Currently, he is working as a Team Lead (Researcher) at AN-DASH Group, SEECS, NUST, Islamabad. His research interests include
computer networks, emerging issues in IEEE 802.11 WLANs, machine learning and social networks.
Zawar Shah completed his PhD degree in Electrical Engineering from the University of New South Wales (UNSW), Sydney, Australia in 2009. Currently, he is
a Senior Lecturer in Information Technology (IT) at Whitireia Community Polytechnic, Auckland, New Zealand. His research interests include QoS issues in
Wireless Networks, Vertical Handover issues between 3G/4G networks, Cloud Computing, Network Architectures and Protocols.
Adeel Baig received PhD and MEngSc degree in computer engineering from the University of New South Wales, Sydney, Australia, in 2007 and 2001,
respectively. Currently, he is an Assistant Professor at School of Electrical Engineering and Computer Science (SEECS), Islamabad. His research interests are
in the protocols and applications for on-board mobile communication networks, network optimization, and QoS provisioning.

Improving QoS of IPTV and VoIP Over IEEE 802.11n

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Computers and Electrical Engineering xxx (2014) xxx–xxx

Contents lists available at ScienceDirect

Computers and Electrical Engineering

Improving QoS of IPTV and VoIP over IEEE 802.11n q

2.1. Transport layer protocols

2.1.1. User Datagram Protocol (UDP)

2.1.2. Datagram Congestion Control Protocol (DCCP)

2.1.3. TCP Friendly Rate Control (TFRC) protocol

2.1.4. TCP Friendly Multicast Congestion Control (TFMCC) protocol

3. Modelling of IPTV and VoIP over IEEE 802.11n

3.1. Network scenario

3.2. Data rate estimation of IPTV and VoIP

Fig. 1. Network scenario for IPTV and VoIP.

Parameter Value Parameter Value

TV F (fps) C Resolution RH RV Compression scheme Required rate (Mbps)

MPEG-2 and H.263 Compression ratio (Hcomp ) = 30:1.

MPEG-4 and H.264 Compression ratio (Hcomp ) = 50:1.

4. Optimal transport layer protocol for IPTV and VoIP

T s ¼ T frame þ SIFS þ T ack þ DIFS ð11Þ

Simulation Analytical Experimental

Simulation Analytical Experimental

Simulation Analytical Experimental

4.3. Fairness analysis of IPTV and VoIP with TCP trafﬁc

Simulation Analytical Experimental

IPTV Combined ﬂows Average throughput (Mbps)

5.1. Trends of SIFS, DIFS and physical header

T DIFS ¼ 2T SIFS þ T Proptime ð17Þ

Parameters Default parameters Proposed parameters [4]

8 SIFS behaviour 7 DIFS behaviour

5.2. Trends of contention window

Fig. 5. Contention window trends for IPTV and VoIP.

5.3. Trends of queue size

5.4. Trends of aggregation

T ¼ T DIFS þ T MPDU þ T SIFS þ T ACK ð23Þ

Fig. 6. Queue size trends for IPTV and VoIP.

Fig. 7. Aggregation trends for IPTV and VoIP.

6. Improving performance through multicast mechanism

6.1. Using multicast for IPTV transmission

Parameters Standard values Simulations [4] Analytical Experimental

Fig. 8. Channel viewership with popularity (Zipf’s Law).

Fig. 9. Network scenario for IPTV and VoIP.

6.2. TFMCC for IPTV transmission

6.3. Wireless enhancement of TFMCC (W-TFMCC)

Terrain TFRC TFMCC Coverage %

Terrain type TFRC TFMCC Coverage %

Algorithm 1. W-TFMCC protocol.

1 pusr ðiÞ; /⁄ Packet loss probability of user i ⁄/

6.4. Test run for W-TFMCC

Fig. 11. Coverage comparison of UDP/TFRC/TFMCC/W-TFMCC.

6.5. Comparison of TFMCC vs W-TFMCC

7. Prior state-of-the-art approaches

[1] Cisco. Cisco visual networking index forecast 2011–2016 <http://goo.gl/hSJjX>.

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