hand-off and synchronization protocols for supporting ... - UPV

HAND-OFF AND SYNCHRONIZATION PROTOCOLS FOR SUPPORTING MULTIMEDIA COMMUNICATIONS IN AN ATM BASED WIRELESS NETWORK Antonio León, Manuel Esteve, Juan C.Guerri, Carlos Palau, Ana Pajares Departamento de Comunicaciones Universidad Politécnica de Valencia Camino de Vera S/N Valencia 46022 SPAIN {aleon,mesteve,jcguerri,cpalau,apajares}@dcom.upv.es ABSTRACT. New multimedia services provision, that requires high transfer speeds, and end to end network guaranteed delays, is an under research field in mobile communications. These services are currently supported by ATM networks, that are, in fact the base of the fixed network in whole mobile communication systems. However, new generations of digital wireless communication systems, based in TDMA, have bandwidth and signalling limitations to support this services. A important problem in this scenario is the synchronisation on multimedia flows when a hand-off is performed. We propose, from a typical architecture ATM fixed network/TDMA wireless system, some improvements in the synchronisation process of multimedia flows after a handoff procedure.

1. INTRODUCTION. Provision of new services requires medium-high speeds. It is the case of audio and digital video, teleconference, high definition images and multimedia applications in general. These are services that need considerable bandwidth and a fixed network can provide today with ATM technology. By other hand, second generation wireless digital systems based on TDMA have several disadvantages in order to provide these multimedia services. Then a valid and logic solution could be a wireless network based on ATM in order to harmonise networks and technologies and obtain easy interconnection. The utilisation of an ATM network has a major drawback, it is the signalling protocol, which is not suited to support users mobility, nor the effects of this kind of scenario. The first problem to be solved is the hand-off process, in which a call is transferred from one cell to

Provision of new services requires medium-high speeds. It is the case of audio and digital video, teleconference, high definition images and multimedia applications in general. These are services that need considerable bandwidth and a fixed network can provide today with ATM technology. By other hand, second generation wireless digital systems based on TDMA have several disadvantages in order to provide these multimedia services. Then a valid and logic solution could be a wireless network based on ATM in order to harmonise networks and technologies and obtain easy interconnection. The utilisation of an ATM network has a major drawback, it is the signalling protocol, which is not suited to support users mobility, nor the effects of this kind of scenario. The first problem to be solved is the hand-off process, in which a call is transferred from one cell to another, through the ATM network without interrupting the communication. Some solutions to this problem have been proposed, another, through the ATM network without interrupting the communication. Some solutions to this problem have been proposed, trying to solve that signalling information does not allow re-routing an established call [1]. Some schemes state that the solution is the modification of the signalling protocol with the addition of new commands and processes that cover users mobility. And there are others who propose a new layer in the protocol stack in charge of the development of these new functions. According to these criteria a more detailed classification can be done: • Cell forwarding [2]. • Dynamic re-routing[1],[3]. • Virtual-tree based [4]. • Full re-establishment. The full re-establishment scheme of the connection requires that a new connection is built-up between the

mobile host and the fixed end-point each time a hand-off is performed. This method is slow and inefficient, although directly developable. The dynamic re-routing scheme only requires that a part of the connection is builtup, while the rest remains unalterable and is profited. This scheme improves to the maximum the channel utilisation, speeding up the hand-off procedure. As drawback the present protocols do not allow this re-routing process once the connection has been established this is why the proposed schemes are based in the addition of signalling messages to the protocols. Regarding the multicast re-routing, it is fast but not efficient debt to the possible duplication of the cells. The main advantage of these schemes is that they can be implemented with UNI 4.0 standard. Finally, it is the re-routing by cells addressing. It is based in the existence of permanent virtual circuits between the stations in order to assist the hand-off procedure. The main advantage is that it keeps the sequence ordering of the cells because all of them follow the same path. This method is useful mainly in plain or ring networks, because in hierarchic networks is easier to re-route than to redirect. Also, in scenarios with small areas (microcells) a queue of cells between the zones would be created, while partial re-directing minimizes always the re-routing. In this paper a new hand-off protocol/procedure is proposed, based in a simple multicast routing algorithm. This protocol allows the implementation of the hand-off procedure of a mobile host in a wireless network without modifying current signalling information. A network architecture is defined similar to other ones in the literature based in zones[5]. These zones are connected by means of an ATM fixed network. The problems associated with multimedia communications in wireless environments, grow if compared with those of a fixed network environments, it is debt to the radio channel bandwidth limitation and the necessity of re-assignation of the available resources when a hand-off procedure appears or new calls are generated. Definitely all these aspects affect the synchronisation of different media flows in a multimedia communication. In the present work, a synchronisation procedure has been added to the hand-off process trying to avoid the probability of packet duplication and loss. This solution is perfect for environments with guarantied QoS and RF channels with enough bandwidth for multimedia information transmission. As the complexity of the systems grow with the addition of delays and the possibility of not having enough bandwidth in the RFC channel, it will be mandatory the introduction of flow control and resources control mechanisms depending on the temporal relationships and available bandwidth. The paper is structured as follows, first of all network architecture considered in the work is presented, after that

the re-routing mechanism and the hand-off protocol, and finishing with a description of the synchronisation process.

2. NETWORK ARCHITECTURE. The Wireless ATM network for which we propose a new hand-off procedure, is going to be composed by mobile terminals or hosts, radio stations and interfaces with the ATM network. The radio stations attached to the same interface will be called a Zone, and a process that will be called base station or BS will manage all of them. We will suppose that the BS in the scenario are connected by means of an ATM network, but the connection between the radio stations and the BSs is not specified. Each terminal can keep several connections at the same time through the wireless ATM network. These connections should be re-routed when a hand-off happens. This process can be studied, dividing the procedure in two stages, the first one, the radio stage in which the radio link is exchanged between radio stations, and the second stage which will support the first one, doing a re-routing and storage of information through the ATM network.

Fig.1: Structure of the Platform Then using the zones concept, the network could be modelled as an architecture with two different levels. We have an inferior layer composed by individual cells and an upper layer composed by a group of individual cells, conforming a zone. This architecture is equivalent to the PLMN (Public Land Mobile Networks) that are formed by Base Stations, BSC (Base Station Controller) y MSC (Mobile Switching Centres). The MSCs execute the switching and signalling tasks that are mandatory for the establishment of the calls originated or terminated in the mobile stations, and where are located the HLR (Home Location Register) and VLR (Visitors Location Register).

Different re-routing schemes can be applied to this network architecture. In a system based in the zones concept, an intra-zone hand-off will be produced when the mobile host changes its position between radio-station or cells of the same zone, it will involve only a re-routing inside the interface equipment of the BS, but not in the ATM network, this kind of hand-off procedures are Mobile Host

Data sent by BS1

BS1

extremely fast, because the network does not take part in the process. But in inter-zone hand-off procedures, the change of position is done between radio stations of different zones, so a re-routing through the nodes of the network would have to be done.

BS2 Invocation

Start

Multicast tree connection time

Done

redirect ACK

Down link Interruption

Data sent by BS2

ACK

ATM Network

Wireless Network

Fig. 2: Hand-off Protocol packet exchange

3. RE-ROUTING MECHANISMS. In the previous section it has been stated that in inter-zone hand-off procedures, re-routing of the connection is basic in order to achieve a satisfactory hand-off procedure [3]. Re-routing the connection consists in the establishment of a new connection starting in a common node of the previously existing one, this node is known as Cross Node. Using this technique a part of the virtual connecting existing previous to the hand-off is kept. The result is faster hand-offs, which are mandatory needed in multimedia applications in microcellular networks. There are different re-routing schemes, based on this principle, of profiting the old connection, to build up a new one . In this paper we are going to introduce another one based on the multicast communication paradigm, which is translated in an easier procedure. In our work, in order to simplify the procedure we have avoided the consistency features (registering of the mobile host in the hone data-bases), because we consider that it has been deeply studied in some previous work [6]. A part from the hand-off procedure, a synchronisation mechanism has been developed, because it is needed if

the application has to ensure about not receiving duplicated data packets. This procedure will allow us to send a data flow besides video and audio. The complete exchange of packets between the elements involved in the hand-off procedure is represented in figure 2. The decision of the kind of handoff (intra or inter-zone) is a BS1 task. If it is an intra-zone hand-off, the cross node is the zone administration node, so no process has to be called. In the contrary, an interzone hand-off a complete exchange of messages to achieve a zone change has to be performed. Following Figure 2, the hand-off procedure works in the following steps: 1. Mobile Host connection is done by means of opening a multicast connection with the information server and the receiving base station, in this case BS1. 2. Mobile Hosts periodically receives control signals form different base stations through a control channel. Then the mobile host can advance the hand-off by studying the power of the different

signals, that is what we call hand-off with prediction. The Mobile Host, after deciding that it needs a hand-off, will send a “start” signal to his actual base station (BS1), reporting the address of the station with which it wants to get connected. 3. As result of this signal, the base station (BS1), establish an unicast connection with the new or destination base station, and sends an “invocation” message with the QoS parameters of the connection. The reason to establish such a connection is to emulate traffic of specific control packets for the hand-off, which still are not available in ATM UNI 4.0 standard. 4. The implementation of the connection partial rerouting is done by connecting to the multicast tree. Until this point only BS1 is sending data to the Mobile Host. But when BS2 is ready to relieve BS1 in the connection it informs BS1 using the unicast connection previously established by BS1, and a “redirect” message, which contains also information about the assigned radio channel, QoS parameters, etc. 5. BS1 receives the transference message, which contains the first data packet received by BS2, if this data packet has been received by BS1, it stops immediately storing information to be sent to the Mobile Host, but if BS1 has not received this packet it waits until the arrival and then stops storing data packets. 6. Following this stage, BS1 informs the Mobile Host by means of a “done” message”, sent through the radio channel. 7. Once the Mobile Host has been reported BS1 leaves the multicast connection, what is equivalent to consider that partial re-routing has been completely and correctly finished. If data transmission was in both directions, before closing it will send all the data through the up link, inform BS2 that it has sent the last packet from the Mobile host and after that it would close the connection. 8. The Mobile Host has received the message that orders the change of station and it reports it to BS2, indicating the last received packet. Since this instant in time the data in the up link will be sent using the new Base Station. 9. When BS2 receives the acknowledge message from the Mobile Host, it start sending the data which had stored since the last one received by the Mobile Host. The hand-off procedure is finished.

The data of the up link are stored until BS1 sends a packet reporting that it has finished its buffer transmission, in order to avoid that some packets could be disordered. This process could be slightly simplified if the multicast tree was permanently established. Nevertheless this model wastes resources, because it obliges to keep opened connections which could be never used, and it also needs information about possible routes of the Mobile Host. It is also important to study the behaviour of this protocol in real-time applications in which a continuous data flow is needed. At first, in the proposed scheme the time needed to achieve the incorporation of the new station does not affects to data flow delays, because the connection is kept by BS1, nevertheless this time influences the total amount of time needed to carry out the hand-off procedure, and it must be considered in microcells hand-offs, in which the overlap stripe between cells is small and so is the permanence time. Another possible solution could be the allowance to BS2 to begin transmission directly after being ready to start, without waiting for a message of the Mobile Host (which will be attached to this base station when it consider appropriate). This solution is easier in terms of packet traffic, but has problems with the continuity of the data flow and some packet lose can appear. When BS2 connects or it starts receiving data, its buffer is empty, and it will directly start transmitting. Nevertheless BS1 has its connection with the Mobile Host still active, and it is sending data from the server, which temporary stores in a buffer in order that the Mobile Host receives it at a correct rate. So if the Mobile Host changes abruptly from BS1 to BS2 maybe some data could be lost because BS1 still has not sent them, but BS2 has already sent them.

4. DESCRIPTION OF SYNCHRONISATION PROCESS.

THE

The hand-off procedure that we have presented in the previous section has been thought to operate in a mobile multimedia application environment so it has to be complemented with a synchronisation protocol, that we are going to describe in this section. The main problem of multimedia communications in wireless environments grows if it is compared with a fixed network environment, because the limitations in the bandwidth in the radio channel, the mandatory need of reassigning the available resources when the hand-off procedure is executed or new calls are generated. All this features influence the multimedia flows synchronisation. The basic synchronisation process in which there are only audio and video, and connected with the hand-off procedure could be implemented in the following steps: 1. Frames sent from the receiver, have a sequence number.

2. When BS2 connects to the multicast tree, and is ready to relieve BS1, it issues a message indicating this fact. In this message BS2 could report the sequence number corresponding to the first frame received of each data flow. 3. When BS1 receives this message it checks if it has been received by itself. If the answer is affirmative (sqBS1>sqBS2) it keeps transmitting and stops receiving more frames. On the contrary, it will keep receiving until (sqBS1=sqBS2). In addition, BS1 would report the Mobile Host in the “done” message about sqBS1. 4. When the Mobile Host receives a “done” message, it checks the value of sqBS1, and when it will receive that frame, it will send an ACK message to BS2, ordering it to start the transmission of all the data in the interval starting in sqBS1+1. Then we can guarantee that there is no duplicates or data loss in the Mobile Host. By other side, it will be needed that the switching time between BSs could be small enough (or the buffer space large enough) in order not to empty the buffer. This solution is valid in environments in which QoS could be guaranteed and the RF channel has enough available bandwidth for multimedia data transmission. But as the complexity of the system increases because of delays, or there are not enough bandwidth available in the RF channel, it will be mandatory to introduce a flow control and resources reallocation mechanisms, depending on the temporal relationships and the available bandwidth. By other side if audio and video were multiplexed and then both were sent using the same virtual circuit, the synchronisation process will be simplified, as happens with MPEG. And if the multimedia flow is composed by different kind of data like audio, video, text and images, where the sequencing of the frames has not relationship among them, synchronisation points could substitute them.

5.GESTION DEL QOS DURANTE EL INTRA AND INTER ZONE HAND-OFF In multimedia applications is evident the importance of providing QoS guarantees. Providing QoS guarantees in wireless networks is a much more complex problem than in fixed networks. The QoS parameters that have to be taken into account in both environments are not the same. In the following, it will be assumed that the fixed network has enough resources to provide QoS guarantees, and that the base stations (BS) have large buffers to solve the jitter problem owing to the fixed network and avoid the transmission rate difference between both networks.

5.1. GESTION DEL QOS DURANTE EL INTRA ZONE HAND-OFF This section describes the structure of the dynamic channel allocation algorithm proposed in this paper. The BS is in charge of running it each time an input event takes place. Figure 2 outlines the block diagram for this algorithm.

State Information

New Call Handoff

Admission Call Algorithm

Call Acceptation and Channel Reervation

Connections State

Reject Connection

Fig 3. Block diagram In this system, two types of input events can be distinguished. • New call arrival: when a new mobile host joins the system, indicate the traffic characterisation it expects to receive, i.e., the bandwidth requirements. This information could correspond with the values specified in [5] and [6] although applications could indicate different values. The BS stores this data in an internal structure (Connections State, Figure 2) that will be used to allocate resources for this connection. Mobile hosts may specify this initial information by using an RTCP packet of type SDES. • Hand-off process: this is a typical mobile system event and takes place when a mobile host changes of cell. The information that the BS needs about the traffic pattern is the same that in the previous case but here, it is exchanged between the base stations (inter-zone) or not (intra-zone). Mobile systems do not need to supply again the traffic characterisation they expect to receive. In this work, new call arrivals and hand-off processes are inputs to the same algorithm. Admission Call Algorithm deals with QoS(BW) or bandwidth guarantees. In the following sections, the algorithm will be discussed with more detail. In a wireless environment, it would be desirable that mobile multimedia applications could accommodate its behaviour to network conditions. This issue is discussed in ¡Error! No se encuentra el origen de la referencia., which deals with adaptive video applications in wireless networks and proposes a layer source coder

that splits up a video signal into its CBR and VBR components. At first, only the CBR portion is transmitted but, in case there is available bandwidth, the VBR component is also transmitted to improve the quality of the image. Adaptive applications are also used in [10], which presents an adaptive framework for QoS guarantees in multimedia wireless networks. This paper considers as well adaptive multimedia applications. Mobile hosts are required to provide a

Audio Real-Time Video Non-Real-Time

Data

characterisation of the data traffic they expect to receive so that the Base Station (BS) will be able to allocate the required resources. In this work, it is distinguished between Real-Time and Non-Real-Time traffic. Table I and Table II show traffic characterisation for each of these traffic patterns. Similar classifications can be found in [8] for bandwidth requirements and [7] for the error rate.

High Quality Audio Voice Videoconference Video On Demand Interactive Data Non Interactive Data

The Real Time Protocol (RTP) ¡Error! No se encuentra el origen de la referencia., is a protocol designed to support adaptive multimedia applications in the Internet network. In RTP, each application source adds a header to the data packet with a sequence number and a timestamp. In addition, each session member, both senders and receivers, sends periodically control packets to every participant of the session by using the Real Time Control Protocol, RTCP. In this way, each source sends sender reports (RTCP SR) which indicate the total number of packets and octets sent, while receivers send receiver reports (RTCP RR) to specify the jitter, loss ratio and the higher sequence number received. In ¡Error! No se encuentra el origen de la referencia. it is shown how to use RTP/RTCP with adaptive applications to detect and avoid congestion by decreasing the application data transmission rate. As ¡Error! No se encuentra el origen de la referencia. suggests, a RTP mixer could be used to reduce the data bandwidth in slow links. In ¡Error! No se encuentra el origen de la referencia., RTP limitations in mobile communication networks are considered and it proposes to locate this mixer at the border of the fixed and the wireless network. The goal of this algorithm is channel allocation to accept or reject new call arrivals depending on the state of the existing connections. So, accepted calls may suffer from variable degrees of QoS(BW) degradation to allow new calls join the system. A new call is admitted if it can get enough available channels to guarantee its minimum QoS(BW). First of all, the search takes place between the available

Minimum Transmission Rate

Peak Rate

QoS(BW)Pri ority

16 Kbps 16 Kbps 256 Kbps 1’2 Mbps 64 Kbps 5 Kbps

192 Kbps --------10 Mbps 2 Mbps 20 Kbps

1 2 3 4

channels. If there are enough channels to guarantee a minimum QoS(BW), the new call is accepted. If it not should be so, the left channels, i.e., the required channels less the available ones, will be look for between the accepted calls. Then, the existing connections could impair its QoS(BW) to let new calls join the system. In [13] two techniques to reduce multimedia application transmission rate are considered. The first one is recoding data in a lower transmission rate, i.e., MPEG-1 video signal could be recode to teleconferencing quality. The second one is selectively discarding data, for example, by throwing out the P or B MPEG-1 frames and delivering the I frames. The candidate connections to lose some channels are those with the highest QoS(BW), and the channels subtracted would be those carrying additional information to increase this QoS(BW). These connections are chosen in accordance with the priority order specified in Table I. Thus, the first candidates to decrease the number of assigned channels are the Non Interactive Data Connections, followed by Interactive Data connections and so on. In those traffic classes that could have several levels of QoS(BW), the best QoS(BW) connections will be chosen the first. The connection would be rejected if it is not possible to find channels for accepting it with a minimum QoS(BW) guarantees. Finally, if the call is accepted and there are still some available channels, they will be used to increase the QoS(BW) of the existing calls. Now, the candidates to improve its QoS(BW) would be those with higher priority, starting by High Quality Audio, continuing on with Video On Demand and so on. In each traffic class, it will be chosen first those connections whose QoS(BW) is the worst one.

When a connection finish, a new channel reassignment is produced to share the released channels between the existing connections. This reassignment aims to allow the maximum number of accepted calls and that all connections may have the maximum QoS(BW) depending on the connection state.

connections to improve its QoS(BW). In addition, when a connection finish, its channels are also reallocated between the existing calls. Therefore, the utilization mean is about 30’68% in the first case while it reaches 95’72% for the second one. QoS (BW)

In order to evaluate the QoS guarantees obtained with the proposed algorithms, a simulation model has been designed. This simulation model has been implemented in C, using SMPL libraries, and has been run in a Sun Sparc Station.

Maximum

New Call Arrival Finalization Call

In the following, simulation results for the proposed algorithm are presented. The results have been gathered in two groups, one of them deals with bandwidth allocation while the other one is related to the error rate.

Minimum

First, the Admission Call Algorithm is evaluated. As it has been described in previous sections, this algorithm is concerned about the QoS(BW) parameter. In order to evaluate its performance, two scenarios have been taken into account. The first one considers that there is a mixer at the BS as is shown in [13]. This approach reduces fixed network bandwidth requirements for accommodating multimedia applications in the wireless slow links. In this case, the BS assigns a fixed number of channels to each connection, which corresponds with a minimum QoS(BW) guarantee, and this assignation remains during the whole communication session. The second scenario uses the dynamic resource allocation algorithm that has been proposed in this paper.

. In Figure 8 , it has been taken an interval of time with six active High Quality Audio connections. When each of them joins the system (New Call Arrival), it is assigned the maximum possible QoS(BW) level, depending on the connection state at that moment. It must be noticed that the behaviour of the existing calls depends on all the active connections and not only on that ones pertaining to their traffic class. As other connections of any class finish (Finalization Call),and due to High Quality Audio calls have the highest priority (see Table I), the on going connections QoS(BW) is enhanced until they reach their peak rate.

0

10

20

Time (s)

Fig 5. High Quality Audio QoS(BW) assignation

0,8

5.1. GESTION DEL QOS DURANTE EL INTER ZONE HAND-OFF

Dynamic QoS(BW)

Si el hando-off producido fuera intrazona el mecanismo sería similar pero en este caso solo se vería implicada la BS correspondiente y, si fuera necesario, el servidor para acomodar la tasa de datos a enviar.

0,6 0,4

9500

10000

9000

8500

8000

7500

7000

6500

6000

5500

5000

4500

3500

3000

2500

2000

1500

1000

0

0

4000

Fixed QoS(BW)

0,2

500

Channel Utilization, U

1

Time (s)

Fig 4. Bandwidth utilization Figure 7 shows bandwidth utilization for each of these approaches. In this figure, it can be noticed a significant gain in bandwidth utilization by using the dynamic algorithm. This is due to, although at first this algorithm assigns to each connection the same amount of bandwidth than the previous approach, afterwards available channels are shared between the accepted

When an inter-zone hand-off is performed, BS1 sends an invocation message. This message indicates the present QoS values and the minimum acceptable by the application. BS2 is in charge of learning the available radio resources in that instant of time. Si los recursos disponibles permiten mantener los valores actuales se lo indicaría a BS1 en el mensaje redirect. Si los recursos permiten unos parámetros QOS diferrentes de los actuales pero superiores al mínimo se lo indicaría a BS1 igual que en el caso anterior pero además enviaría previamente un mensaje al servidor para que acomodara la información enviada a las nuevas restricciones del enlace radio.

Por último si BS2 no puede proporcionar los valores Qos mínimos rechazaría la conexión en el mensaje de respuesta.

[5] D.C. Cox , ‘Wireless Network Access for Personal [6]

Hay que observar que en este capítulo nos hemos centrados en los problemas de lograr ciertos valores QoS debido a las limitqaciones en el enlace radio. Los problemas de mantener los parametros en la red fija al incorporar nuevas estaciones al árbol multicast se escapan al contenido de este artículo.

6. CONCLUSIONS. When a mobile station moves from one to next zone, a rerouting process must be implemented. In addition, to support multimedia services, flow synchronisation is required to maintain quality of service. We have proposed a re-routing and a synchronisation procedures to implement these actions in an ATM/wireless environment.

7. REFERENCES. [1] [2] [3]

[4]

B. A. Akyol, D. C. Cox. ‘Signalling Alternatives in a Wireless ATM Network’. IEEE J.S.A.C. Special Issue on Wireless ATM. January 1997. R. Yuan, S. K. Biswas, D. Raychaudhuri. ‘A Signalling and Control Architecture for Mobility Support in Wireless ATM Networks’. ICC’ 96. C-K Toh. ‘A Hybrid Hand-over Protocol For Wireless ATM LANs’. ACM Journal on Mobile Networks & Applications-Special Issue on ‘Wireless ATM’ ACM/Baltzer Press. December 1996. A. S. Acampora. ‘Wireless ATM: A Perspective on Issues abd Prospects’. IEEE Personal Communications. August 1996.

Communications’, IEEE Comm. Mag., Vol.30, No 12, pp. 96-115, December 1992. A. S. Acampora, M. Naghshineh, ‘An Architecture and Methodology for Mobile-Executed Hand-off in Cellular ATM Networks’. IEEE J.S.A.C. October 1994.

[1] R. Braden, D. Clark, S. Shenker. “Integrated Services in the Internet Architecture: an Overview”. Request for Comments 1633, Internet Engineering Task Force, June 1994. [2] L. Zhang, S. Deering, D. Estrin, S. Shenker, D. Zappala. “RSVP: A New Resource Reservation Protocol”. IEEE Networks, September 1993. [3] H. Adiseshu, G.Parulkar. “A State Setup Protocol”. Technical Report, Washington University, St.Louis, 1997. [4] M. Inoue, G. Wu, Y. Hase. “Channel State Dependent Resource Scheduling For Wireless Message Transport”. IEEE, Vehicular Tecnology Conference Proceedings. pp. 1264-1268, 1998. [5] M. Woo, N. Prabhu, A. Ghafoor. “Dynamic Resource Allocation for Multimedia Services in Mobile Communication Environments”. IEEE Journal on Selected Areas in Communications, vol 13, No 5, pp. 913-922, June 1995. [6] C. Oliveira, J. Bae Kim, T. Suda. “Quality-of-Service Guarantee in High-Speed Multimedia Wireless Networks”. IEEE International Conference on Communications, vol 2, pp. 728-734, 1996 [7] K. Brown, S. Singh. “Extensions to RTP to support Mobile Networking”. Technical Report, University of South Carolina, 1997. [8] H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications”. RFC1889. January 1996. [9] M. Esteve, J.C Guerri, C.Palau, V.Casares. “Flow Control Mechanism with Hysteresial Techniques”. IEE Electronic Letters, vol 34, No5, pp332-333. February 1998.