EVALUATION OF ACHIEVED VIDEO QUALITY IN WIRELESS
MULTIMEDIA TRANSMISSION USING UDP-LITE
Janne Vehkaperä, Johannes Peltola, Jyrki Huusko, Mikko Myllyniemi, and Mikko Majanen
VTT Technical Research Centre of Finland
Kaitoväylä 1, Oulu, P.O. Box 1100, FIN-90571
Finland
firstname.lastname@vtt.fi
ABSTRACT
The purpose of this paper is to study the performance of
the partial checksum mechanism of UDP-Lite transport
protocol compared to the standard UDP protocol in
wireless environment in terms of achieved video quality.
Video quality is evaluated in PSNR using simulation
results achieved with a simulation platform developed in
the framework of IST PHOENIX project. The platform
offers the necessary components for modeling all relevant
ISO/OSI layers for wireless IP video streaming, including
partial checksum functionality on transport and data link
layers. The achieved gain in video quality in terms of
PSNR for IEEE 802.11a simulation model validates that
the video streaming benefits of using UDP-Lite transport
protocol instead of using regular UDP in wireless
channels. The enhancement was 2 – 3 dB in PSNR for
AWGN channel and for fading channels the difference
was smaller, being 1dB or less. The major conclusion of
our study is that the best improvement in video quality
using UDP-Lite is achieved in conditions where bit errors
appear in very short bursts. Also a minor enhancement
using UDP-Lite is achieved in transmission conditions,
where signal-to-noise ratio in the channel is very low.
KEY WORDS
Wireless multimedia, transport layer, UDP-Lite, video
quality, and MPEG-4.
1. Introduction
The new generation of wireless networks enable more
efficient transmission of multimedia content to mobile
terminals and video communication is becoming more
and more popular. However, the transmission over
wireless links stays challenging due to varying
transmission conditions and high transmission error rates.
Novel multimedia applications can handle corrupted
packets and they often even prefer corrupted packets
instead of lost packets. However, erroneous data is
usually not delivered to the application because it is
discarded by checksum process performed at lower layers
including data link and transport layer checksum
mechanisms. Sending high bit rate and delay-sensitive
multimedia stream through wireless networks requires
support from the application, transport and link layers for
allowing corrupted data to be passed to higher layers and
utilizing the available bandwidth more efficiently.
User Datagram Protocol (UDP) [4] is a best-effort
transport protocol and it has become very popular in
delay-sensitive multimedia applications due to its
simplicity. However, UDP does not pass corrupted
packets; it simply discards them. UDP-Lite [2] is a
modified version of the UDP transport protocol
introduced by IETF and it has been developed to allow
delivery of corrupted packets to higher layers by
introducing more lightweight checksum calculation.
Suitability of UDP-Lite for multimedia transmission has
been studied in several papers [1], [3], [6] and [10].
However, in-depth analysis of the effect of UDP-Lite to
the achieved video quality compared to UDP has not been
made. Studies have been concentrated on better
throughput provided by UDP-Lite and the achieved video
quality is not deeply analyzed using common video
quality metrics.
In this paper, we study the performance of the partial
checksum mechanism proposed by UDP-Lite compared to
the standard UDP protocol in terms of achieved video
quality. To compare the performance of these protocols,
we have made several simulations utilizing a software
platform developed in the framework of the IST
PHOENIX project (www.ist-phoenix.org) [5]. We
evaluate the performance of UDP-Lite using different
wireless channel models namely additive white Gaussian
noise (AWGN), fast fading, and frequency non-selective
block fading channel models.
This paper is organized as follows. Section 2 introduces
the system architecture for the optimized wireless
multimedia transmission and the proposed solutions for
different layers. Section 3 introduces the developed
simulation model and Section 4 presents simulation
results achieved utilizing the simulation model. Section 5
summarizes some key conclusions of this paper.
2. System Architecture and Protocol Stack
The Figure 1 represents the protocol architecture for user
datagram protocol (UDP) and UDP-Lite performance
analysis. The system consist of transmitter (source coder)
and receiver (source decoder) units. Data streams are
packetized with Real-Time Transport Protocol (RTP) and
the data stream is delivered through IPv6 network. The
model consist the wired IPv6 network. The last hop in the
system is considered wireless. The Section 3 of this paper
describes the wireless channel and modelled protocols in
more detail.
Figure 1. Protocol architecture for multimedia
transmission
2.1 Transport layer
The function of the transport layer protocols is to provide
end-to-end communication between two (or more) hosts.
For multimedia streaming in IP networks, current de-facto
standard is to use RTP together with UDP.
RTP [7] provides (among other things) timestamps and
sequence numbers to packets. Sequence numbers can be
used for detecting lost packets or in sequencing of the out-
of-order packets. Timestamps are used to maintain
synchronisation with the media time and the play back
time. RTP also provides payload type identification. The
companion control protocol, RTCP, monitors the quality
of service (among other things). It may, for example,
inform the sender to slow down the transmission rate due
to the detected increase in packet loss rate.
UDP [4] provides connectionless, best-effort, maybe-
duplicates, unordered transport service. It is quite a simple
protocol and has a low overhead. It adds demultiplexing
and (optional) data integrity service (checksum) to IP.
Using UDP checksum is optional with IPv4 but
mandatory with IPv6 because IPv6 does not have any
checksum mechanism of its own. Also, disabling UDP
checksum in IPv4 is not reasonable since damaged
transport and higher layer headers might pass unverified
and the packets might be delivered incorrectly. UDP has
neither error reporting nor error recovery mechanism:
damaged packets are simply discarded.
UDP-Lite [2] introduces the partial checksum to UDP. If
errors are detected in the sensitive part (e.g. in the header)
of the packet, it will be discarded. If errors are in the
insensitive part of the packet (e.g. in the payload), the
packet is not discarded. The sensitive part of the packet
always includes at least the IP pseudo header and the
UDP header. Additionally, the sensitive part can be
extended to cover also parts of the payload, e.g. RTP
header. Thus, UDP-Lite allows the data integrity checking
in vital parts of the packets, thus prohibiting all kind of
misbehaviours due to the corrupted header information,
but on the other hand, it allows errors in less important
parts of the packet. It has been shown that the number of
discarded packets is significantly reduced when using
UDP-Lite instead of UDP [3]. This is something that error
resilient video decoder may utilise for better video quality
after error concealment tools.
The partial checksum is achieved by changing the UDP’s
Length field to Coverage field – it specifies the coverage
of the checksum and it is defined by the sending
application on a per-packet basis. This change is possible
because some redundancy is included in headers (UDP
length field = IP length field – size of IP header). By
setting Coverage value equal to packet length turns UDP-
Lite into traditional UDP. The UDP-Lite header together
with IPv6 pseudo header is depicted in the Figure 2.
Figure 2. IPv6 pseudo header and UDP-Lite header
2.2 Data Link Layer
In order to utilize UDP-Lite effectively in multimedia
delivery, the data link layer should also support the
transmission of corrupted data. Two different solutions
can be applied for transmitting the corrupted data to upper
layers at medium access control (MAC) as described in
[1]. The solutions are; to ignore the checksum at MAC
level, or to provide the partial checksum for MAC
transmission frames.
Ignoring totally the checksum mechanism for
transmission frame is inadvisable since it can lead to the
situation, where the system crashes due to the erroneous
packet header information and faulty protocol
interpretation. Thus, partial checksum mechanism is
preferred also for MAC level.
We have defined a novel partial checksum solution for the
carrier sense medium access with collision avoidance
(CSMA/CA) type of medium access, which is used in our
simulation model. Our mechanism differs slightly from
some previously published partial checksum mechanism
such as one described in [8].
The MAC frame format for multimedia transmission is
illustrated in Figure 3. The proposed solution enhances
the IEEE 802.11 standard in multimedia delivery. In the
IEEE 802.11a/b/g based systems, the multimedia data
frame is identified with frame control field, and the
mechanism is backward compatible with original
standard. Our mechanism is based on fast byte-by-byte
cyclic redundancy check (CRC) operations instead of bit-
by-bit hardware operations, which are utilized for
example in majority of WLAN hardware solutions. The
benefits of fast byte-by-byte software solution are
represented e.g. in [9] and with state-of-the-art hardware
solutions the proposed partial checksum mechanism is
feasible. Including the easy CRC calculation and per
frame checking routines, our MAC frame structure
benefits from the dynamic CRC coverage area for
different data frames and streams.
• IPv6 network protocol,
• IPv6 network and mobility modeling,
• Robust Header Compression (RoHC),
• Enhanced MAC protocol,
• Physical layer controller,
• Channel en/decoder (convolution, RCPC, and
low density parity check (LDPC) codes),
• Interleaver and modulator (OFDM, TCM,
TTCM, STTC),
• Radio channel model (AWGN, fast and slow
fading channels).
Data lines:

Payload
Source Cross layer communication Source
decoding Block specific control decoding

Streaming Streaming
protocol protocol
e.g. RTP & End to end QoS e.g. RTP &
RTCP RTCP

Figure 3. MAC frame format for multimedia data Transport Transport

with partial checksum support protocol
e.g. UDPLite
protocol
e.g. UDPLite

IP
IP IP IP
network

3. Simulation Model Data link control Data link control

and Medium and Medium
Access Control Access Control

The simulation framework in this paper utilizes a

Channel
simulation platform developed in EU-IST funded decoding and
demodulation
Channel
decoding and
demodulation
PHOENIX project. The primary purpose of this
simulation tool is to allow development of different cross-
layer optimization techniques for wireless video
Sending IP
streaming communication systems. The simulation terminal Radio link transmitter Receiving wireless terminal

platform contains different models for all system layers
from the physical layer up to the application layer present
in IP based video communication networks. Accurate
modeling of the physical layer functionality and channel
as well as seamless integration of the application layer
tools such as controllers and video codecs distinct this
simulation platform from traditional network simulation
tools such as Opnet or ns-2. Modeling of the physical
layer allows obtaining realistic simulation results for
techniques, such as UDP-Lite where generated error
patterns in the channel have strong effect on the
transmission performance.
As stated earlier, all of the system layers have been
implemented into the simulation platform in order to
provide a realistic evaluation of the system performance
[5]. System model for the simulation environment is
illustrated in Figure 4. Following system layers are
included in the platform and can be used in the
simulations:
Figure 4. System model of the simulation environment
The most important blocks affecting into the simulations
and parameters used within each block in this study are
introduced in more detail in the following chapters.
3.3 Source codec
We have used MPEG-4 simple profile video coding
standard for video compression. MPEG-4 standard
includes several techniques, such as resynchronization
markers, data partitioning, reversible variable length
codes (RVLC), and header extension codes (HEC) which
improve both packet and bit error resilience of the
compressed video. The decoder includes additional error
concealment methods such as spatial error concealment,
where correctly received image areas are used to predict
lost image areas to further improve perceived video
quality.

• Application layer controller, 3.4 Transport layer

• Source codec (MPEG-4,H.264, and H.264/SVC),
• Ciphering,
• Unequal error protection (UEP) at application
layer using rate compatible punctured
convolution (RCPC) coding,
• RTP packetization and RTCP signaling,
• Transport layer (UDP/UDP-Lite/Datagram
congestion control protocol (DCCP),
The bit stream produced by the source encoder is further
packetized using RTP protocol, which adds timestamps
and sequence numbers into the packet in order to improve
the transmission above best-effort transport protocols.
After RTP packetization, UDP or UDP-Lite protocol is
used to achieve low-delay transmission of video content.
In the UDP-Lite protocol implementation, a checksum
calculation can be defined to cover IP pseudo header and
UDP-Lite header or to cover also a defined portion of the
payload in addition to these headers. In UDP-Lite
simulations, the checksum is calculated over pseudo IP,
UDP-Lite and RTP headers. In UDP simulations, the
checksum is calculated using the whole packet (headers
and payload).
3.5 IPv6 network
The simulation platform includes a simulated wired IPv6
network where topology of the network and delay
introduced by the network can be defined. In our
simulations, the effect of wired network is minimized in
order to test the performance of the UDP-Lite protocol in
the last wireless hop. This means that the delays
introduced by the wired IPv6 network are short and no
packet losses occur in wired network.
3.6 Data link layer
An enhanced IEEE 802.11 MAC layer is assumed, since
no fragmentation or packing is considered for
transmission frames. Partial CRC is performed for each
data frame using the mechanism defined in Section 2.
Automatic retransmissions are not considered since only a
small delay introduced by retransmission cycle for MAC
data frames is assumed. Simulations represent the worst
case scenario, where data frames are discarded after the
predefined retransmission cycle.
3.7 Radio channel
The platform includes a full physical layer
implementation for generic radio interface including
channel models for AWGN, fast and slow fading
channels. Radio model includes different interleaving,
channel coding and modulation schemes. For this study
we have used simulation model for IEEE 802.11a using
QPSK modulation. The actual modulation scheme should
not have great impact on PSNR video quality results, thus
results should be also applicable for IEEE 802.11b
systems.
3.8 Cross layer signalling
The simulation platform has been designed to support
cross-layer optimization research for wireless video
streaming. This support is enabled by offering adaptive
system components, cross-layer controllers and dedicated
cross-layer communication control signals across
different layers. Adaptive system components indicate
that all components should be able to change their
parameters or operation mode on demand. Cross layer
controllers collect information about the status of system
layers and based on this information they adjust the
operation of different components. For this work the cross
layer functionality has been used to signal the coverage
area for UDP and data link layers checksum calculation.
4. Simulation Results
In order to evaluate the possible enhancement in video
quality achieved using UDP-Lite instead of UDP, we
have simulated the transmission of MPEG-4 video over
wireless channel using introduced simulation platform.
The scenario considered is a video transmission using
IEEE 802.11a WLAN radio. The maximum bit rate is 12
Mb/s from which 1 Mb/s is considered for simulated
transmission rate since the capacity of the channel is
shared among several users. The transmitter and the
receiver are considered to be inside a building in an office
environment and the transmitter is considered to stay in
one position and the receiver is assumed to move slowly
in a floor area. We have considered three different types
of channel models for our simulations: AWGN, fast
fading, and frequency non-selective block fading
channels. Fading channel models with average SNR of 6
dB are illustrated in Figure 5. QPSK modulation and
RCPC channel coding are used in every simulation. For
each channel type we have calculated the packet loss ratio
(PLR) and PSNR video quality metric for different
channel SNR conditions.
Fast fading channel model (mean SNR=6 dB)
15
SNR [dB]
10
5
0
0 10 20 30 40 50 60
Time [s]
Frequence non-selective block fading
channel model (mean SNR=6.3 dB)
20
SNR [dB]
10
0
-10
0 10 20 30 40 50 60
Time [s]
Figure 5. Fast fading and frequency non-selective
channel (below) models
As a test video sequence, we have used CIF-resolution
Foreman sequence compressed using MPEG-4 at average
bit rate 448 kbit/s. The frame rate of the video is 30 fps
and intra-refresh is made after every 15 frames. Error
resilience tools are used and the video packet size was
2500 bits which means in our simulations that one
predicted frame (P-frame) is typically divided into two
separate packets. In the case of AWGN channel,
simulations have also been done with a video packet size
of 1500 bits in order to study the effect of the packet size
to the video quality. In the following, we will present the
simulation results for each channels type.
4.1 Achieved quality in AWGN channel
PSNR and PLR as a function of channel SNR are shown
for AWGN channel in the Figures 6 and 7. Distinct
difference between UDP-Lite and UDP can be noticed
when the packet size is 2500 bits. Effective PSNR
improvement for UDP-Lite is located at channel SNR
range from 1dB to 3dB. In this region 2 - 3dB
improvement in PSNR was measured. If the channel was
very bad (SNR smaller than 0dB) or very good (SNR
greater than 4), the difference between UDP and UDP-
Lite saturates to zero. Same behaviour can be seen from
PLR results. In the effective range, PLR differ about 30
percentage units, but outside of this region there is no
difference between UDP and UDP-Lite results.
4.2 Achieved quality in fast fading channel
Figure 8 presents simulation results for fast fading
channel, where the sampling period for fast fading gain
was 1 ms and Doppler frequency for time correlated
Rayleigh fading was 5 Hz. These fading parameters
correspond roughly to walking inside the office area.
Peak signal-to-noise ratio in fast fading channel
30
UDP-Lite
25

PSNR [dB]
UDP
20
Peak signal-to-noise ratio in awgn channel 15

30 10
-4 -2 0 2 4 6 8
PSNR

channel snr [dB]

UDP-Lite
20 Packet loss rate in fast fading channel
UDP 1

UDP-Lite
10
0 2 4 6 UDP

PLR
channel snr [dB] 0.5

Packet loss rate in awgn channel

0.9
0
UDP-Lite -4 -2 0 2 4 6 8
0.6 channel snr [dB]
PLR

UDP

0.3 Figure 8. PSNR and PLR curves in fast fading channel

0
0 2 4 6 Achieved video quality in fast fading channel simulation
channel snr [dB] was moderate. The average PSNR was below 25 dB for
Figure 6. PSNR and PLR curves in AWGN channel both UDP and UDP-Lite for all measured channel SNR
for packet size of 2500 bits values. UDP-Lite gave better PSNR values, but the
difference between UDP and UDP-Lite was only about
Peak signal-to-noise ratio in awgn channel 1dB. The small difference in PSNR is partly explained by
with packet size 1500 bits the small difference also in PLR statistics. The results can
30 be explained by the fact that fading causes a long burst or
PSNR [dB]

20
errors which reach the headers as well as payload, thus
UDP-Lite
UDP
UDP-Lite discards most packets affected by error bursts.
10

0 1 2 3 4 4.3 Achieved quality in frequency non-selective block

channel snr [dB]
fading channel
Packet loss ratio in awgn channel
with packet size 1500 bits
1 Figure 9 illustrates simulation results for frequency non-
selective block fading channel with uncorrelated log-
PLR

0.5 UDP-Lite normal distributed slow fading, where is 16 dB and the

UDP
coherence time for slow fading gain is 2000 ms. The
0
0 1 2 3 4
results for this channel model are similar to results for the
channel snr [dB] fast fading channel in terms of differences in PSNR and
Figure 7. PSNR and PLR curves in AWGN channel PLR values between UDP and UDP-Lite transmission.
for packet size of 1500 bits The differences in PSNR are even slightly smaller than in
the case of fast fading channel being mostly under 1dB.
Figure 7 presents AWGN results when the packet size is
1500 bits and thus smaller than in Figure 6. It can be seen
that when a larger packet size is used, UDP-Lite reaps
more benefit compared to UDP. When using a smaller
packet size there are relatively more errors in header parts
than in the payload part and this causes smaller PLR
difference for smaller packets. In addition, errors in the
payload may cause loss for VLC synchronization which
effectively may have similar effect for small payload than
losing short packet altogether.
 Peak signal-to-noise ratio in frequency
non-selective block fading channel
35
PSNR [dB]
30 UDP-Lite
UDP
25
20
15
-2 0 2 4 6
channel snr [dB]
Packet loss rate in frequency
non-selective block fading channel
0.6
0.4
PLR
0.2
UDP-Lite
UDP
0
-2 0 2 4 6
channel snr [dB]
Figure 9. PSNR and PLR curves in frequency non-
selective block fading channel
5. Conclusions
In this paper, we have evaluated the achieved
enhancement in video quality in wireless transmission,
where corrupted packets are allowed to reach the video
decoder instead of dropping packets in the network layers.
We have used both UDP-Lite and modified MAC
protocols which introduce the partial checksum
calculation for packet payload in order to allow passing
corrupted packets to error-resilient video decoder. To
evaluate the achieved video quality in wireless
transmission using UDP-Lite, we introduced the
simulation platform which has been developed in IST-
PHOENIX project. The simulation platform can be used
for evaluating different models and algorithms in all
system layers from the physical layer up to the application
layer present in IP based video communication networks.
We have used this simulation environment to evaluate the
performance of the UDP-Lite protocol using three
different wireless channel models.
Simulations for the considered wireless channel models
conclude that allowing corrupted packets to reach the
error-resilient video decoder will improve the achieved
video quality. The greatest improvement in video quality
was achieved in AWGN channel where bit errors have
more random pattern. In fast fading channel and
especially in frequency non-selective block fading
channel, the improvement in video quality was smaller. In
frequency non-selective block fading channel the bit
errors occur usually in bursts and this will reduce the
quality of the channel to intolerable and significant
improvement in video quality using UDP-Lite instead of
UDP is not achieved. According to our simulations, the
key finding of our study is that the best improvement in
video quality using UDP-Lite wireless transmission is
achieved in transmission conditions where only short
bursts of bit errors occur. In poor transmission conditions
only a minor enhancement in video quality using UDP-
Lite instead of UDP is achieved. A retransmission could
be more effective solution in poor transmission conditions
if the amount of retransmitted data is not high and it does
not affect on the real-time requirements.
Acknowledgements
8 10
This work was carried out in the PHOENIX project (FP6-
2002-IST-1-001812), which was partially funded by the
European Commission within the European Union Sixth
Framework Programme and Information Society
Technologies. The authors would like to thank the whole
project consortium and the colleagues who have
participated in the development of the PHOENIX system.
8 10
References
[1] L-Å. Larzon, M. Degermark, and S. Pink, UDP Lite
for Real Time Multimedia Applications. Proc of the IEEE
International Conference of Communications (ICC),
Vancouver, British Columbia, Canada, June 1999.
[2] L-Å. Larzon, M. Degermark, S. Pink, L-E. Jonsson
(ed.), and G. Fairhurst (ed.), The Lightweight User
Datagram Protocol (UDP-Lite). RFC 3828, July 2004.
[3] R. Reine and G. Fairhurst, MPEG-4 and UDP-lite for
Multimedia Transmission. 4th Annual Post Graduate
Networking Conference, PGNET 2003, Liverpool, UK,
June 2003.
[4] J. Postel, User Datagram Protocol. RFC 768, August,
1980.
[5] M.G. Martini, M. Mazzotti, C. Lamy-Bergot, P.
Amon, G. Panza, J. Huusko, J. Peltola, G. Jeney, G.
Feher, and S.X. Ng, A Demonstration Platform for
Network Aware Joint Optimization of Wireless Video
Transmission. Proceedings of 15th IST Mobile & Wireless
Communication Summit, Myconos, Greece, June 4-8,
2006.
[6] S.A. Khayam, S. Karande, H. Radha, and D.
Loguinov, Performance analysis and modeling of errors
and losses over 802.11b LANs for high-bit-rate real-time
multimedia, Signal Processing: Image Communication,
18(7), 2003, 575-595.
[7] H. Schulzrinne, S. Casner, R. Frederick, and V.
Jacobson, RTP: A Transport Protocol for Real-Time
Applications. RFC 3550, July 2003.
[8] E. Masala, M. Bottero, J.C. De Martin, MAC-Level
Partial Checksum for H.264 Video Transmission over
802.11 Ad Hoc Wireless Networks, in Proc. of IEEE 61st
Semiannual Vehicular Technology Conference (VTC),
Stockholm, Sweden, May-June 2005.
[9]. T. Ritter, The Great CRC Mystery, Dr. Dobb's
Journal of Software Tools, vol 1, number 2, February,
1986, pp. 26-34, 76-83.
[10] A. Singh, A. Konrad, and A.D. Joseph, Performance
evaluation of UDP lite for cellular video. Proc. of the 11th
NOSSDAV, Port Jefferson, New York, USA, June 25-27,
2001.