Priority Based Congestion Control for Multimedia Traffic In 3G Networks

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					                                                        (IJCSIS) International Journal of Computer Science and Information Security,
                                                        Vol. 8, No. 6, September 2010


                       Priority Based Congestion Control for
                        Multimedia Traffic In 3G Networks
                            Neetu Sharma1, Amit Sharma2 , V.S Rathore3, Durgesh Kumar Mishra4
                                        123
                                         Department of Computer Engineering, Rajasthan, India
                                   13
                                      Rajasthan College of Engineering for women, Rajasthan, India
                                  2
                                   Shri Balagi College of Engineering & Technology, Rajasthan, India
                                  4
                                    Acropolis Institute of Technology and Research, Indore, MP, India
                                     neetucom10@gmail.com, amitit_04@rediffmail.com
                                    drvsrathore@rcew.ac.in,drdurgeshmishra@gmail.com


ABSTRACT- There is a growing demand for efficient                       TCP    friendly rate    control   (TFRC) and        Adaptive increase
multimedia streaming applications over the Internet and next            multiplicative decrease (AIMD) used in networks. These algorithms
generation mobile networks. Multimedia streaming services are           used for multimedia traffic but not much effective in packet loss.
receiving considerable interest in the mobile network business.         TCP is the dominant transport protocol in the Internet, and the
As communication technology is being developed, the user                current stability of the Internet depends on its end-to-end congestion
demand for multimedia services raises. The third generation             control, which uses an Additive Increase Multiplicative Decrease
(3G) mobile systems are designed to further enhance the                 (AIMD) algorithm. End-to-end congestion control of best-effort
communication by providing high data rates of the order of 2            traffic is required to avoid the congestion collapse of the global
Mbps. High Speed Downlink Packet Access (HSDPA) is an                   Internet [11]. While TCP congestion control is         appropriate   for
enhancement to 3G networks that supports data rates of several          applications such as bulk data transfer, some real-time applications
Mbit/s, making it suitable for applications like multimedia, in         (that is, where the data is being played out in real-time) find halving
addition to traditional services like voice call. Services like         the sending rate in response to a single congestion indication to be
person-to-person two way video calls or one way video calls, aim        unnecessarily severe, For providing a better congestion control with
to improve person-to-person communication. Entertainment                higher data rates a new effective scheme is used. Congestion control
services like gaming, video streaming of a movie, movie trailers        is an important issue in both wired and wireless streaming
or video clips are also supported in 3G. Many more of such              applications. Multimedia applications should use some form of
services are possible due to the augmented data rates supported         congestion control, both in wired and cellular networks, in order to
by the 3G networks and because of the support for Quality of            adapt the sending rate to the available bandwidth. Today’s Internet
Service (QoS) differentiation in order to efficiently deliver           stability is due to TCP and its congestion control algorithm. TCP
required quality for different types of services.                       represents a very efficient transport protocol in general and is
                                                                        suitable for data transfer. However, it has been argued [13] that TCP
This paper present congestion control schemes that are suitable
                                                                        is unsuitable for video streaming because strict delay and jitter
for multimedia flows. The problem is that packet losses, during
                                                                        requirements of video streaming are not respected by TCP.
bad radio conditions in 3G, not only degrade the multimedia
                                                                        Moreover, some TCP retransmissions are unnecessary for video
quality, but render the current congestion control algorithms as
                                                                        when data may miss the arrival deadline and become obsolete. This
inefficient. This paper proposed a solution that integrated the
                                                                        has led researchers to look for alternative options. Most of the work
congestion control schemes with a priority based multimedia
                                                                        related to congestion control for video flows has either emulated TCP
packets to increase the speed of multimedia data and reduce the
                                                                        or has used the TCP model. The well-known TCP-Friendly Rate
packet loss that is developed due to congestion in networks
                                                                        Control (TFRC) congestion control consists in an equation based rate
Key words: UMTS, CN, BS, TFMCC, UTRAN, RNC
                                                                        control mechanism [13][14][15], designed to keep a relatively steady
                                                                        sending rate while still being responsive to congestion. When used
                  I.        INTRODUCTION
                                                                        over wireless links, TFRC and TCP cannot distinguish between the

The emerging multimedia application requires a fresh approach for       wireless losses and the congestion losses. They both may suffer from

congestion control. A widely popular congestion control schemes are     the link underutilization if the connection traverses a wireless link.




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This is because they consider dropped packets as a sure sign of          UMTS networking architecture is organized in two domains. The
congestion and reduce the ending rate significantly. The inability to    user equipment (UE) and the public land mobile network (PLMN).
identify a wireless loss followed by unnecessary reduction in sending    The UE is used by the subscriber to access the UMTS services.
rate results in link underutilization.                                   PLAN is further divided into two land-based infrastructures
A. UMTS Introduction
                                                                              (i)        UTRAN (UMTS terrestrial radio– access network)
Universal Mobile Telecommunications System (UMTS) is a third-
                                                                              (ii)       CN (core network).
generation (3G), wireless cellular network that uses Wideband Code
Division Multiple Access (WCDMA) as its radio interface                  The UTRAN handless all radio-related functionalities and the CN is
technology. UMTS offers higher data rates with respect to older 2G       responsible for maintaining subscribes data and for connections.
and 2.5G networks and, with the Release 5 version, is evolving into      UTRAN contain two types of nodes Radio network controller (RNC)
an all-IP, wireless network. The increased bandwidth provided by         and Node B. Node B is the base station and provides radio coverage
UMTS allows for the deployment of a wide range of services, like         to one or more cells. Node B connected to UE via Uu interface and to
voice, data and multimedia streaming services. In wireless networks,     the RNC via Iub interface. Uu is a radio interface based on the
congestion control, alone, may not be enough to ensure good quality      wideband code division multiple access (WCDMA) technology [7].
of multimedia streaming and efficient utilization of the network.
                                                                         The CN consist by two types of general packet radio service support
Packet losses due to the high bit error rate not only degrade the
                                                                         nodes (GSNs). That is gateway GSN (GGSN) and serving GSN
multimedia quality, but render the current congestion control
                                                                         (SGSN). SGSN provide the routing functionality. It manages a group
algorithms as inefficient: these algorithms back-off on every packet
                                                                         of RNCs and interacts with the home location register (HLR). HLR
loss even when there is no congestion. We integrate the congestion
                                                                         permanently store the subscriber data. SGSN connected to GGSN via
control schemes with an adaptive retransmission scheme in order to
                                                                         the Gn interface. RNC connect to SGSN via Iu interface. Through the
selectively retransmit some lost multimedia packets. Fig.1 shows the
                                                                         GGSN the UNTS network connect to external packet data network
transmission of multimedia data over a wireless channel.
                                                                         like the internet.




                 Fig. 1 Transmission of Multimedia data                                        Fig.2 General UMTS Network

                                                                         C. 3G/UMTS Problems
B. GENERAL UMTS NETWORK:
                                                                                Problems due to the use of IP
                                                                                         o    IP   doesn’t    support     real   time    streaming
UMTS, the successor of GSM, is evolving toward a future wireless
                                                                                              requirements
all-IP network. In this paper we present how it supports real-time IP
                                                                                         o    Overhead due to packet header
multimedia services, as these services are expected to drive the
                                                                                    Problems due to radio conditions
adoption of wireless all-IP networks.
                                                                                         o    Scarce and time varying bandwidth
                                                                                         o    Congestion, wireless losses & large delay




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D. UMTS QoS Classes                                                        control profiles that may be used with the DCCP transport protocol
UMTS defines four QoS classes [2] and the classified traffic gets the      [10]; TFRC may also be implemented by UDP-based applications
treatment inside the UMTS network according to its class. The four         wishing to perform congestion control. This paper presents a
QoS classes are:                                                           simulation study of TFRC over UMTS networks supporting

• Conversational class: The traffic from the applications like person-     HSDPA. Since we are interested in video streaming applications, we
to-person video or voice call is classified into conversational class.     evaluate the performance of TFRC in terms of rate stability over
The delay and jitter requirements for this type of traffic are very        different time scales, and compare it with that of TCP. Several
strict. This is because on the both end points there is a human            scenarios of MAC-layer scheduling, radio conditions and background
expecting the delivery of the voice and/or video data in very short        traffic are considered.
time after it is sent.
                                                                           This paper proposed a more reliable algorithm that provides
• Streaming class: Video on Demand (VoD) falls under this class.           congestion control for different multimedia classes. Priority assigned
The delay requirements are there but are not as strict as the              to each of the packet according to multimedia classes. So whenever
conversational class.                                                      the congestion occurs in the network the lowest priority packets are
                                                                           dropped. If overall loss rate for lower priority packets is not very
• Interactive class: The interactive traffic like interactive e-mail or
                                                                           high, then we can safely assume that the congestion loss rate for the
web browsing falls under this category. Though there is still some
                                                                           highest priority packets will be insignificant. In such a case, the loss
delay requirement, it is less strict than the conversational and
                                                                           of highest priority packets will be mainly due to wireless errors.
streaming classes. Moreover, since the traffic mostly pertains to data
                                                                           Thus, it is to be expected that, in general, there is a good correlation
applications, the bit error rate should be very low.
                                                                           between wireless packet loss rate and the total loss rate of highest
• Background class: This class is the most insensitive to delay. It        priority packets.
includes the traffic from background applications like background
email and SMS. Though, the bit error rate, like the Interactive class,
                                                                                               2. THE PROPOSED SCHEME

should be very low.                                                        This paper provides a mechanism of congestion control for the

D Congestion Control for Multimedia data                                   multimedia transmission over UMTS. We analyze TCP friendly
                                                                           multicast congestion control (TFMCC) over UMTS and generalize it
TCP-Friendly Rate Control (TFRC) is an end-to-end congestion
                                                                           to different multimedia classes [5][6]. We design a novel mechanism
control mechanism, whose goal is to provide rate control for unicast
                                                                           for congestion control that is Content Sensitive TCP Friendly
flows in IP networks. The main feature of TFRC is its ability to
                                                                           Multicast Congestion Control (CSTFMCC). We perform a little
smoothly adapt the sending rate of a flow to network conditions,
                                                                           modification in UMTS network and the packet field. At various level
while competing for bandwidth with TCP flows in a relatively fair
                                                                           of network we provide the control mechanism that prevents the
manner. TFRC was designed to offer a more stable sending rate than
                                                                           network from the congestion. Multimedia Class I traffic includes
TCP on wired, best-effort networks, making it suitable for
                                                                           video and audio traffic from users equipped with an adjustable rate.
applications like multimedia streaming. We evaluate the performance
                                                                           Class II traffic includes non-real time data traffic such as e-mail, file
of TFRC, compare it with that of TCP and new TFRC for different
                                                                           transfer and web browsing traffic. These two classes contain different
multimedia classes, under different scenarios of MAC-layer
                                                                           multimedia traffic that is more delay sensitive or less delay sensitive.
scheduling, radio conditions and background traffic.
                                                                           So class I traffic support the real time applications and more delay
TFRC [4][10] is an end-to-end congestion control mechanism                 sensitive. Due to congestion, if any loss of the packet or the delay
suitable for applications with constraints on rate stability, like voice   between the packets can reduce the quality of received video/audio.
or streaming media. It has been designed to adapt the sending rate of      Whether in class II traffic, if congestion occur it is acceptable to
a flow in a smooth manner, while trying to fairly share the available      buffer non-real time data at a network node or at the user station and
bandwidth with competing TCP flows. TFRC is an Internet standard           transmit them at a slower rate. In a large multicast group, there will
[4], and it has been adopted at the IETF as one of the congestion          usually be at least one receiver that has experienced a recent packet




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                                                                                                        ISSN 1947-5500
                                                           (IJCSIS) International Journal of Computer Science and Information Security,
                                                           Vol. 8, No. 6, September 2010



loss. If the congestion control mechanisms require that the sender        If (incoming request for higher priority packets)
                                                                                    If (there is a free channel) then
reduces its sending rate in response to each loss, as in TCP, then
                                                                                                allocate the free channel
there is little potential for the construction of scalable multicast                else
                                                                                    If (lower priority packets)
congestion control.
                                                                                                put in a buffer
                                                                                                If (there is free channel again)
In wireless communication systems like UMTS, the packet loss may                                            allocate the free channel to lower
not mean network congestion. The quality of wireless link may be                                priority packets
                                                                                                else
degrading due to signal fading. During a fading period, the bit error                           Ignore request
rate of wireless link may become very high but after that period the                            endif
                                                                                    else
wireless link is expected to recover. TFMCC uses a feed back                        Ignore request
scheme which allows the receiver to calculating the slowest                         endif
                                                                                    ignore request
transmission rate to always reach the sender.                                       endif
                                                                          endif
                                                                          End.
A. Sender end: Fragmentation of data packets perform at the sender
end. The sender fragment the data packet with on bit of priority.                                          Fig.3 Algorithm For proposed model

There are two parts of the data (i) packet header and (ii) payload. The
size of header part change by one bit shows the priority of packet. So    Fig. 4 shows the flow chart for the proposed model. Flow chart
                                                                          shows the arrival of packet and priority check by the routers at
there is only one bit modification perform in the size of data packet     layer2. According to this priority the packets is being processed.
and it increases the speed of multimedia packets.
                                                                                                          Packet Arrival




B. Multimedia Packet Size: Multimedia packet size depends on the                            High          Check priority of
                                                                                                          Packets?
                                                                                                                                 Low




multimedia classes. The proposed scheme redesigns the multimedia               Process the
                                                                               higher priority                                             Process the
                                                                               packets                                                     lower priority
                                                                                                                                           packets
packets. It increases the multimedia packet size by one bit. This bit
shows the priority of multimedia packets. The highest priority                  Any Free           No                                       Any Free            Yes
                                                                                                                                            Channel?
                                                                                Channel?

packets serve first by the routers at the layer 2. So the size of the                   Yes                                                          No

                                                                                                                                  Yes
                                                                                                                                            Buffer full ?
packet is increased by one bit.                                                                                                                    No
                                                                            Accept packet and           Block (reject)        Buffer the                    Accept packet and
                                                                            assign channel              the packet            packet                        assign the channel

C. Routing Scenario: For fast transmission of multimedia
information the proposed scheme give the priority to all multimedia
                                                                                                         Fig. 4 Flow Chart for the proposed model
packets. When a user want to send multimedia data the data framing
perform at the sender end. The sender constructs the frame with a
priority bit. This information stored in the header of the packet for     Receiving End: At the receiving end defragmentation perform. The
priority access to the router. Sender sends the packets towards its       receiving data packet reaches at the destination and multimedia
destination. Multimedia packets reach at the network. At layer two        information is available for the user respectively
the router checks the destination address and priority bit of the
                                                                          A. Simulation Platform
packet. If a higher priority packet arrives then router serves first to
the packet which contains a highest priority. This increases the speed    The simulation that we use for this is EURANE (NS-2
of multimedia packets and decreases the congestion in the networks.       Extension)[16]. Following fig. 3 shows the simulation topology to
                                                                          increase the multimedia quality. This paper focuses on the problem
For implementing this scenario the changes perform in the size of
                                                                          of evaluating the subjective video quality and presents the quality
packets and in routers. Following algorithm shows the scenario for
                                                                          estimation tool that we employed. a performance evaluation study
routing the various packets according to priority. Fig. 3 shows the
                                                                          done with the well known ns-2 network simulator
algorithm for the proposed model.




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                                                                                 [6] Ljiljana    Trajkovic      and    S.     Jamaloddin    Golestant”
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                          Fig:3 Simulation Topology
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                                                                                      Viho*     and Gábor Jeney.
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data packet. User equipment (UE) has some additional function to                      IEEE IEEE "GLOBECOM" 2008

detect the multimedia class type. The additional functionality of UE             [12] S. Floyd and K. Fall. Promoting the Use of End-to-end

and the field of data packet has main target to remove the packet loss                Congestion     Control      in    the    Internet.    IEEE/ACM

and congestion.                                                                       Transactions on Networking, Aug. 1999.
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             INFOCOM 2004 Hong Kong China                                             ID 3: TCP-Friendly Rate Control (TFRC)., RFC 4342,
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Authors Profile                                          Architecture, Operating System Fundamentals,
                                                         DBMS & RDBMS (Oracle, DB2, SQL Server, MS-
Ms. Neetu Sharma, Reader                                 Access,     DBASE,      etc.),   Data  Structures,
                                                         Programming Languages (C, C++, Java (J2SE,
                                                         J2ME, J2EE), VB, COBOL), Networking
                                                         Technologies (Data Communications, Internet &
                                                         Intranet,    E-Commerce,       Network   Security,
                                                         Cryptology etc.), Software Engineering, System
                                                         Analysis & Design, Management Information
                                                         System, Decision Support System, Artificial
                                                         Intelligence, E-Governance, Computer Center
                                                         Management, UNIX, etc.. He is the member of
Biography: Mrs. Neetu Sharma obtained her                renowned society like ISTE.
Engineering degree from University of Rajasthan
and Masters Degree from Rajasthan Vidyapeeth,            Dr. Durgesh Kumar Mishra
Udiapur securing First division with honors in both.     Professor (CSE) and Dean (R&D),
Currently she is pursuing Ph.D. (CSE) in                 Acropolis Institute of Technology and Research,
Congestion Control in 3G from Gyanvihar                  Indore, MP, India,
                                                         Ph - +91 9826047547, +91-731-4730038
University, Jaipur, India. She has been Reader and       Email: durgeshmishra@ieee.org
HOD of the department of CSE at Rajasthan
College of Engineering for Women, Jaipur, India.         Chairman IEEE Computer Society, Bombay Chapter
She has extensively worked in various field of           Vice Chairman IEEE MP Subsection
Computer Engineering. She has published many
national papers in the reputed journals and
conferences. She is an author of the book 'System
Software Engineering' for B.Tech. students. She is
the member of renowned societies like IEEE, IEEE
computer society, ISTE and CSI also.


Dr. Vijay Rathore, Associate Professor


                                                         Biography: Dr. Durgesh Kumar Mishra has
                                                         received M.Tech. degree in Computer Science from
                                                         DAVV, Indore in 1994 and PhD degree in
                                                         Computer Engineering in 2008. Presently he is
                                                         working as Professor (CSE) and Dean (R&D) in
                                                         Acropolis Institute of Technology and Research,
                                                         Indore, MP, India. He is having around 21 Yrs of
                                                         teaching experience and more than 7 Yrs of
                                                         research experience. He has completed his research
Biography: Dr. Vijay Singh Rahore obtained his           work with Dr. M. Chandwani, Director, IET-DAVV
MCA and Ph.D. (CSE) from University of                   Indore, MP, India in Secure Multi- Party
Rajasthan, India. He is an Associate Professor,          Computation. He has published more than 60
Shree Karni College, Jaipur, India. He has more          papers in refereed International/National Journal
than 10 years of industrial and teaching experience.     and Conference including IEEE, ACM etc. He is a
His areas of interest are Computer Organization &        Senior Member of IEEE, Chairman of IEEE




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                                                                                      ISSN 1947-5500
                                          (IJCSIS) International Journal of Computer Science and Information Security,
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Computer Society, Bombay Chapter, India. Dr.
Mishra has delivered his tutorials in IEEE
International conferences in India as well as other
countries also. He is also the programme committee
member of several International conferences. He
visited and delivered his invited talk in Taiwan,
Bangladesh, Nepal, Malaysia, Bali-Indonesia,
Singapore, Sri Lanka, USA and UK etc in Secure
Multi-Party Computation of Information Security.
He is an author of one book also. He is also the
reviewer of tree International Journal of
Information Security. He is a Chief Editor of
Journal of Technology and Engineering Sciences.
He has been a consultant to industries and
Government organization like Sale tax and Labor
Department of Government of Madhya Pradesh,
India.

Mr. Amit Sharma, Assistant Professor




Biography: Mr. Amit sharma obtained his MCA
from University of Rajasthan and aboout to
complete his M.Tech.(CSE) from Rajasthan
Technical University. He is an Assistant Professor
in Sri Balaji College of Engineering & Technology,
Jaipur. He has more than 5 years of industrial and
teaching experience. His areas of interest are Open
Source,      Networking,     Advance      Computer
Architecture and Information Security. He is the
member of renowned society like ISTE.




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