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Priority Based Congestion Control for Multimedia Traffic In 3G Networks

Priority Based Congestion Control for Multimedia Traffic In 3G Networks

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Published by ijcsis
Abstract- There is a growing demand for efficient multimedia streaming applications over the Internet and next generation mobile networks. Multimedia streaming services are receiving considerable interest in the mobile network business. As communication technology is being developed, the user demand for multimedia services raises. The third generation (3G) mobile systems are designed to further enhance the communication by providing high data rates of the order of 2 Mbps. High Speed Downlink Packet Access (HSDPA) is an enhancement to 3G networks that supports data rates of several Mbit/s, making it suitable for applications like multimedia, in addition to traditional services like voice call. Services like person-to-person two way video calls or one way video calls, aim to improve person-to-person communication. Entertainment services like gaming, video streaming of a movie, movie trailers or video clips are also supported in 3G. Many more of such services are possible due to the augmented data rates supported by the 3G networks and because of the support for Quality of Service (QoS) differentiation in order to efficiently deliver required quality for different types of services. This paper present congestion control schemes that are suitable for multimedia flows. The problem is that packet losses, during bad radio conditions in 3G, not only degrade the multimedia quality, but render the current congestion control algorithms as inefficient. This paper proposed a solution that integrated the congestion control schemes with a priority based multimedia packets to increase the speed of multimedia data and reduce the packet loss that is developed due to congestion in networks

Keywords: UMTS, CN, BS, TFMCC, UTRAN, RNC
Abstract- There is a growing demand for efficient multimedia streaming applications over the Internet and next generation mobile networks. Multimedia streaming services are receiving considerable interest in the mobile network business. As communication technology is being developed, the user demand for multimedia services raises. The third generation (3G) mobile systems are designed to further enhance the communication by providing high data rates of the order of 2 Mbps. High Speed Downlink Packet Access (HSDPA) is an enhancement to 3G networks that supports data rates of several Mbit/s, making it suitable for applications like multimedia, in addition to traditional services like voice call. Services like person-to-person two way video calls or one way video calls, aim to improve person-to-person communication. Entertainment services like gaming, video streaming of a movie, movie trailers or video clips are also supported in 3G. Many more of such services are possible due to the augmented data rates supported by the 3G networks and because of the support for Quality of Service (QoS) differentiation in order to efficiently deliver required quality for different types of services. This paper present congestion control schemes that are suitable for multimedia flows. The problem is that packet losses, during bad radio conditions in 3G, not only degrade the multimedia quality, but render the current congestion control algorithms as inefficient. This paper proposed a solution that integrated the congestion control schemes with a priority based multimedia packets to increase the speed of multimedia data and reduce the packet loss that is developed due to congestion in networks

Keywords: UMTS, CN, BS, TFMCC, UTRAN, RNC

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Published by: ijcsis on Oct 10, 2010
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11/03/2012

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 ABSTRACT 
-
 
There is a growing demand for efficientmultimedia streaming applications over the Internet and nextgeneration mobile networks. Multimedia streaming services arereceiving considerable interest in the mobile network business.As communication technology is being developed, the userdemand for multimedia services raises. The third generation(3G) mobile systems are designed to further enhance thecommunication by providing high data rates of the order of 2Mbps. High Speed Downlink Packet Access (HSDPA) is anenhancement to 3G networks that supports data rates of severalMbit/s, making it suitable for applications like multimedia, inaddition to traditional services like voice call. Services likeperson-to-person two way video calls or one way video calls, aimto improve person-to-person communication. Entertainmentservices like gaming, video streaming of a movie, movie trailersor video clips are also supported in 3G. Many more of suchservices are possible due to the augmented data rates supportedby the 3G networks and because of the support for Quality of Service (QoS) differentiation in order to efficiently deliverrequired quality for different types of services.This paper present congestion control schemes that are suitablefor multimedia flows. The problem is that packet losses, duringbad radio conditions in 3G, not only degrade the multimediaquality, but render the current congestion control algorithms asinefficient. This paper proposed a solution that integrated thecongestion control schemes with a priority based multimediapackets to increase the speed of multimedia data and reduce thepacket loss that is developed due to congestion in networks
Key words: UMTS, CN, BS, TFMCC, UTRAN, RNC
I. INTRODUCTION
The emerging multimedia application requires a fresh approach forcongestion control. A widely popular congestion control schemes areTCP friendly rate control (TFRC) and Adaptive increasemultiplicative decrease (AIMD) used in networks. These algorithmsused for multimedia traffic but not much effective in packet loss.TCP is the dominant transport protocol in the Internet, and thecurrent stability of the Internet depends on its end-to-end congestioncontrol, which uses an Additive Increase Multiplicative Decrease(AIMD) algorithm. End-to-end congestion control of best-efforttraffic is required to avoid the congestion collapse of the globalInternet [11]. While TCP congestion control is appropriate forapplications such as bulk data transfer, some real-time applications(that is, where the data is being played out in real-time) find halvingthe sending rate in response to a single congestion indication to beunnecessarily severe, For providing a better congestion control withhigher data rates a new effective scheme is used. Congestion controlis an important issue in both wired and wireless streamingapplications. Multimedia applications should use some form of congestion control, both in wired and cellular networks, in order toadapt the sending rate to the available bandwidth. Today
s Internetstability is due to TCP and its congestion control algorithm. TCPrepresents a very efficient transport protocol in general and issuitable for data transfer. However, it has been argued [13] that TCPis unsuitable for video streaming because strict delay and jitterrequirements of video streaming are not respected by TCP.Moreover, some TCP retransmissions are unnecessary for videowhen data may miss the arrival deadline and become obsolete. Thishas led researchers to look for alternative options. Most of the workrelated to congestion control for video flows has either emulated TCPor has used the TCP model. The well-known TCP-Friendly RateControl (TFRC) congestion control consists in an equation based ratecontrol mechanism [13][14][15], designed to keep a relatively steadysending rate while still being responsive to congestion. When
 
usedover wireless links, TFRC and TCP cannot distinguish between thewireless losses and the congestion losses. They both may suffer fromthe link underutilization if the connection traverses a wireless link.
Priority Based Congestion Control forMultimedia Traffic In 3G Networks
Neetu Sharma
1
, Amit Sharma
2
,V.S Rathore
3
,Durgesh Kumar Mishra
4
 
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.comdrvsrathore@rcew.ac.in,drdurgeshmishra@gmail.com
(IJCSIS) International Journal of Computer Science and Information Security,Vol. 8, No. 6, September 2010167http://sites.google.com/site/ijcsis/ISSN 1947-5500
 
This is because they consider dropped packets as a sure sign of congestion and reduce the ending rate significantly. The inability toidentify a wireless loss followed by unnecessary reduction in sendingrate results in link underutilization.
 A. UMTS Introduction
Universal Mobile Telecommunications System (UMTS) is a third-generation (3G), wireless cellular network that uses Wideband CodeDivision Multiple Access (WCDMA) as its radio interfacetechnology. UMTS offers higher data rates with respect to older 2Gand 2.5G networks and, with the Release 5 version, is evolving intoan all-IP, wireless network. The increased bandwidth provided byUMTS allows for the deployment of a wide range of services, likevoice, data and multimedia streaming services. In wireless networks,congestion control, alone, may not be enough to ensure good qualityof multimedia streaming and efficient utilization of the network.Packet losses due to the high bit error rate not only degrade themultimedia quality, but render the current congestion controlalgorithms as inefficient: these algorithms back-off on every packetloss even when there is no congestion. We integrate the congestioncontrol schemes with an adaptive retransmission scheme in order toselectively retransmit some lost multimedia packets. Fig.1 shows thetransmission of multimedia data over a wireless channel.
Fig. 1 Transmission of Multimedia data
 B. GENERAL UMTS NETWORK 
:
UMTS, the successor of GSM, is evolving toward a future wirelessall-IP network. In this paper we present how it supports real-time IPmultimedia services, as these services are expected to drive theadoption of wireless all-IP networks.UMTS networking architecture is organized in two domains. Theuser equipment (UE) and the public land mobile network (PLMN).The UE is used by the subscriber to access the UMTS services.PLAN is further divided into two land-based infrastructures(i)
 
UTRAN (UMTS terrestrial radio
 – 
access network)(ii)
 
CN (core network).The UTRAN handless all radio-related functionalities and the CN isresponsible for maintaining subscribes data and for connections.UTRAN contain two types of nodes Radio network controller (RNC)and Node B. Node B is the base station and provides radio coverageto one or more cells. Node B connected to UE via Uu interface and tothe RNC via Iub interface. Uu is a radio interface based on thewideband code division multiple access (WCDMA) technology [7].The CN consist by two types of general packet radio service supportnodes (GSNs). That is gateway GSN (GGSN) and serving GSN(SGSN). SGSN provide the routing functionality. It manages a groupof RNCs and interacts with the home location register (HLR). HLRpermanently store the subscriber data. SGSN connected to GGSN viathe Gn interface. RNC connect to SGSN via Iu interface. Through theGGSN the UNTS network connect to external packet data networklike the internet.
Fig.2 General UMTS Network
C. 3G/UMTS Problems
 
Problems due to the use of IP
o
 
IP doesn’t support real time streaming
requirements
o
 
Overhead due to packet header
 
Problems due to radio conditions
o
 
Scarce and time varying bandwidth
o
 
Congestion, wireless losses & large delay
(IJCSIS) International Journal of Computer Science and Information Security,Vol. 8, No. 6, September 2010168http://sites.google.com/site/ijcsis/ISSN 1947-5500
 
 D. UMTS QoS Classes
UMTS defines four QoS classes [2] and the classified traffic gets thetreatment inside the UMTS network according to its class. The fourQoS classes are:
Conversational class
: The traffic from the applications like person-to-person video or voice call is classified into conversational class.The delay and jitter requirements for this type of traffic are verystrict. This is because on the both end points there is a humanexpecting the delivery of the voice and/or video data in very shorttime after it is sent.
Streaming class
: Video on Demand (VoD) falls under this class.The delay requirements are there but are not as strict as theconversational class.
Interactive class
: The interactive traffic like interactive e-mail orweb browsing falls under this category. Though there is still somedelay requirement, it is less strict than the conversational andstreaming classes. Moreover, since the traffic mostly pertains to dataapplications, the bit error rate should be very low.
Background class
: This class is the most insensitive to delay. Itincludes the traffic from background applications like backgroundemail and SMS. Though, the bit error rate, like the Interactive class,should be very low.
 D Congestion Control for Multimedia data
TCP-Friendly Rate Control (TFRC) is an end-to-end congestioncontrol mechanism, whose goal is to provide rate control for unicastflows in IP networks. The main feature of TFRC is its ability tosmoothly adapt the sending rate of a flow to network conditions,while competing for bandwidth with TCP flows in a relatively fairmanner. TFRC was designed to offer a more stable sending rate thanTCP on wired, best-effort networks, making it suitable forapplications like multimedia streaming. We evaluate the performanceof TFRC, compare it with that of TCP and new TFRC for differentmultimedia classes, under different scenarios of MAC-layerscheduling, radio conditions and background traffic.TFRC [4][10] is an end-to-end congestion control mechanismsuitable for applications with constraints on rate stability, like voiceor streaming media. It has been designed to adapt the sending rate of a flow in a smooth manner, while trying to fairly share the availablebandwidth with competing TCP flows. TFRC is an Internet standard[4], and it has been adopted at the IETF as one of the congestioncontrol profiles that may be used with the DCCP transport protocol[10]; TFRC may also be implemented by UDP-based applicationswishing to perform congestion control. This paper presents asimulation study of TFRC over UMTS networks supportingHSDPA. Since we are interested in video streaming applications, weevaluate the performance of TFRC in terms of rate stability overdifferent time scales, and compare it with that of TCP. Severalscenarios of MAC-layer scheduling, radio conditions and backgroundtraffic are considered.This paper proposed a more reliable algorithm that providescongestion control for different multimedia classes. Priority assignedto each of the packet according to multimedia classes. So wheneverthe congestion occurs in the network the lowest priority packets aredropped. If overall loss rate for lower priority packets is not veryhigh, then we can safely assume that the congestion loss rate for thehighest priority packets will be insignificant. In such a case, the lossof highest priority packets will be mainly due to wireless errors.Thus, it is to be expected that, in general, there is a good correlationbetween wireless packet loss rate and the total loss rate of highestpriority packets.
2. THE PROPOSED SCHEME
This paper provides a mechanism of congestion control for themultimedia transmission over UMTS. We analyze TCP friendlymulticast congestion control (TFMCC) over UMTS and generalize itto different multimedia classes [5][6]. We design a novel mechanismfor congestion control that is Content Sensitive TCP FriendlyMulticast Congestion Control (CSTFMCC). We perform a littlemodification in UMTS network and the packet field. At various levelof network we provide the control mechanism that prevents thenetwork from the congestion. Multimedia Class I traffic includesvideo and audio traffic from users equipped with an adjustable rate.Class II traffic includes non-real time data traffic such as e-mail, filetransfer and web browsing traffic. These two classes contain differentmultimedia traffic that is more delay sensitive or less delay sensitive.So class I traffic support the real time applications and more delaysensitive. Due to congestion, if any loss of the packet or the delaybetween the packets can reduce the quality of received video/audio.Whether in class II traffic, if congestion occur it is acceptable tobuffer non-real time data at a network node or at the user station andtransmit them at a slower rate. In a large multicast group, there willusually be at least one receiver that has experienced a recent packet
(IJCSIS) International Journal of Computer Science and Information Security,Vol. 8, No. 6, September 2010169http://sites.google.com/site/ijcsis/ISSN 1947-5500

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