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Compressed Video Communications Abdul Sadka Copyright © 2002 John Wiley & Sons Ltd ISBNs: 0-470-84312-8 (Hardback); 0-470-84671-2 (Electronic) 5 Video Communications Over Mobile IP Networks 5.1 Introduction The near future will witness the universal deployment of the third-generation mobile access networks that are expected to revolutionise the world of telecom-munications. In addition to conventional voice communication services provided by the second-generation GSM networks, the third-generation mobile networks will support a greatly enhanced range of services due to the higher throughput made available by embracing a number of new access technologies. These include TDMA and a variety of CDMA radio access families such as the direct sequence Wideband-CDMA(WCDMA)and multi-carrierCDMA. Consequently,the most prominent development brought forward by the third-generation family of stan-dards and protocols, namely IMT-2000, compared to second-generation GSM systems, is the provision of high data rates that will enable the support of a wide range of real-time mobile multimedia services including combinations of video, speech/audio and data/text traffic streams with QoS control (Third-generation Partnershipproject). This chapter examines the issues involved in the provisionof videoservices over the 2.5G and 3G mobile networks, and evaluates the perceived service quality resulting from video transmissions over these networks under various operating conditions. The focus will also be on describing and analysing the performance of a number of tools specifically designed to improve the percep-tual video quality over the new mobile access networks. 5.2 Evolution of 3GMobile Networks The second-generation GSM technology has resulted in a major success for the delivery of telephony and low bit rate data services to mobile end users. On the other hand,the tremendousgrowthof the Internethas given rise to a new rangeof multimedia applications that have penetrated the global market at an explosive 178 VIDEO COMMUNICATIONS OVER MOBILE IP NETWORKS pace. The aim of the third-generation mobile networks is to combine the multi-media services of the Internet and the digital cellular concept of mobile radio networks in order to support the provision of multimedia services over mobile wireless platforms. In order to accommodate a new range of services with much higher data rates thanthoseprovidedby GSM,the mostfundamentalimprovementthat isrequired from the third-generation mobile systems is to embrace a number of new access technologies that will allow for a high-throughput access and true real-time multimedia services. The fundamental voice communication services provided by the 2G GSM will be preserved by the new mobile systems, while assuring an improvedaudioqualityacross thenetworkalongwithimprovedcallmanagement and multiparty communication. In addition to conventional voice services, the mobile users will have the ability to connect to the Internet remotely while retaining access to all its facilities, such as e-mail and Web browsing sessions. Mobile terminals will be enabled to access remote websites and multimedia-rich databaseswith the use of multimediaplug-insembeddedinto the Web browsersof these terminals. The conversational video communicationsover 3G networks will also support multi-user capabilities such as multi-party videoconferencingamong various fixed and mobile users. The ubiquity of connection that is allowed by portable mobile terminals will significantly enhance the functionalities of such devices, especially in scenarios involving e-commerce and e-business applications. This will be made possible through the implementation of mobile work environ-ments and virtual offices. Last but not least, the next generation of mobile networks will also support the selective and on-demand coverage of live events such as breaking news and sports in the form of streaming audiovisual content. Thiswill also be accompaniedby the on-demandaccess to archived media such as high-quality highlights of TV scenes and remote audiovisual clips. Thesupport to allthe mobilemultimediaservicesmentionedabovewill haveits implicationsfor the design of the end-to-endmobile network architecture. Firstly, the quality of service (QoS) offered to client applications will be a function of differentconnectionparameterssuchasthroughput,end-to-enddelays,errorrates and frame dropping rates. Therefore, each mobile terminal will have access to a number of bearer channels, each offering a different QoS to the various services being used. On the other hand, the standardised protocols that were adopted for the Internet Protocol and have consequently led to the widespread success of the Internet have allowed an extremely diverse range of terminals and devices to communicate with each other. Moreover, the accepted application-layer stan-dards such as the HyperText Transfer Protocol (HTTP) have also allowed multi-media applications to be deployed and to proliferate. The combination and interoperability of these universally accepted application and network-layer stan-dards will certainly constitute the core of the architecture of 3G systems, and will identify the mechanism of operation of multimedia services over these mobile platforms. This chapter will focus on the real-time transmission of compressed 5.3 VIDEO COMMUNICATIONS FROM A NETWORK PERSPECTIVE 179 Figure 5.1 Evolution of mobile networks video data encapsulated in IP packets over the future mobile networks.Figure 5.1 illustrates the time evolution of mobile networks as a function of their provided services. This evolution was consolidated by the remarkable migration from the second-generationGSM network to the third-generationEDGE (Enhanced Data rate GSM Evolution) and UMTS (Universal Mobile TelecommunicationSystem) networks through the 2.5G packet-switching GPRS (General Packet Radio Ser-vice) and circuit-switching HSCSD(High Speed Circuit Switched Data) systems. 5.3 Video Communications from a Network Perspective One of the main design trends of multimedia networks is to achieve a connection between two or more users by bringing digital content, such as video, to their desktops. Video telephony, videoconferencing, telemedicine and distance learning are all examples of multimedia applications that aim at providing video (along withvoice)servicesinanetworkingenvironment.Beyondthedesktop,multimedia technology relies on high-capacity digital networks to carry video content and support real-time services such as messaging, conversation, live and on-demand streaming, etc. In video telephony and conferencing for instance, users are geo-graphically far from each other and therefore the video streams must be transmit-ted in real time over a communicationnetwork. In video on-demandapplications, thestoragemediumis remote,andthus videomustbe retrievedandstreamedover a network for being delivered to the requesting user. In distance learning applica-tions, video is captured and then transmitted to remote learners using a shared communicationmedium.Inallthesecases,acommunicationnetworkis obviously required. Since the users are located far from each other, multimedia services must be offeredinthe presenceofatelecommunicationsystemthat performsthe routingof 180 VIDEO COMMUNICATIONS OVER MOBILE IP NETWORKS multimedia traffic across a network. On the other hand, a multimedia service might involve more than two users at the same time (such as videoconferencing). This requires the presence of a sophisticated network infrastructure with an integral communication protocol for the end-to-end routing, transport and deliv-eryofmultimediatraffic.Withoutthedevelopmentofcorporatenetworksto route the video traffic among various users, little chance exists to commercialise multi-media and broaden its applications from the PC-based software and hardware to multi-sharing services on a worldwide basis. 5.3.1 Why packet video? The time synchronisation between the sender and receiver is a key issue in any communicationsession.Toachievesynchronisation,either oneof twoapproaches is adopted, namely synchronous and asynchronous transmissions. Asynchronous communicationconsistsof sending the streamof data in the formof symbols, each representedbya pre-definednumberofbits.Eachsymbolisprecededby astartbit and followed by a parity bit, thereby leading to an overhead of two bits per symbol. With synchronous transmission, characters are transmitted without any start and end indicators. However, to enable the receiver to determine the begin-ningand end of a block of data (set of characters),each block of data begins with a preamble bit pattern and ends with a post-amble bit pattern, as is the case in asynchronous communication systems. This block of data is referred to as a packet. The packet can be of fixed length such as the ATM cell (53 bytes), or variable length as for IP packets. Unlike data streams, coded video has a very low tolerance to delay, and therefore dropped video information cannot be retransmitted.Alternatively, com-pressed video data has to be fitted into a certain structure that enables error control to be applied in case of information loss and bit errors. This structure is calleda packetandconsistsof avideopayloadand aprotocolheader. Theprocess of fitting the video payload into this packet structure is called packetisation, and the part of the communication system where packetisation is performed is known as the packetiser. Figure 5.2 is a block diagram of a typical packetiser with one input video source. A number of advantages are obtained from packetising a compressed video stream before transmission. It is intended that a number of applications would be running between two end-points at the same time. Moreover, the traffic flow between these two end-points may consist of a number of various traffic types. Therefore, the successful end-to-end control and delivery of routed multimedia information would be impossibleiftheinformationbitswerenotsentinpacketformat.Thetraffictypeof thepayloadisthenidentifiedbythe contentofthe typefieldineachpacketheader. Using the packet structure, it would be possible to multiplex various streams of 5.3 VIDEO COMMUNICATIONS FROM A NETWORK PERSPECTIVE 181 Figure 5.2 Block diagram of a video packetiser/depacketiser system dataonto thesame bearersincethe depacketiserwouldthen beable toidentifythe source of each packet from the content of its type field. Once the source is known, the payload is then delivered to the corresponding decoder. Consequently, the packet structure enables the multiplexing of various streams of data, thereby resulting in an efficient sharing of the available bandwidth. Duetoexcessivedelaysandinterference,thevideodataissubjecttoinformation loss and bit errors, respectively. As examined in Chapter 4, a single bit error could lead to a disastrous degradation of the decoded video quality. If a packetisation scheme is employed, the effect of bit errors and information loss could be confined to a single packet since the video decoder would then resynchronise at the beginning of the following error-free packet. Moreover, the MBs contained in a video packet can be predicted independently of the MBs in other packets (Inde-pendent Segment Decoding in Annex R of H.263] described in Section 4.12), thereby improving the error robustness of video data. The packet structure enables the datagram or connectionless service of the network layer routing protocol. As opposed to the virtual circuit connection, the connectionlessrouting strategy shows a high flexibility in the selection of the path between source and destination at any instant of time. It also results in a much higher channel utilisation, since it does not require any prior bandwidth alloca-tion, as is the case for virtual circuit connections. To prevent out-of-sequence arrival of packets, resulting from multipath fading and varying network condi-tions, the depacketiser can re-order the received packets in accordance with their sequence numbers before passing their payload up to the video decoder. One further advantage of packet transmission is the ability of the decoder to acknowledge the receipt of error-free packets. In many situations, it is paramount that the video encoder is aware of the network conditions so that it adapts its output rate and error protection mechanism accordingly. The acknowledgement of correct delivery can be periodically sent to the encoder in the form of feedback ... - tailieumienphi.vn