Maximum bitrate for uplink Maximum bitrate for downlink 20, 40, 72 kbps

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1 An Enhanced Quality Of Service Method for Guaranteed rate Services over Shared Channels in EGPRS Systems Daniel Fernández and Héctor Montes Nokia Networks, C/ Severo Ochoa, s/n Edif. de Inst. Universitarios, Pl. 3., Parque Tecnológico de Andalucía Campanillas (Málaga) Spain ext-daniel.fernandez@nokia.com, ext-hector.montes@nokia.com Abstract- The goal of this paper is to present a complete Quality Of Service (QoS) framework for the Radio Resource Management (RRM) able to provide guaranteed bitrate (GBR) services, when they are transmitted over shared channels in a Enhanced General Packet Radio Service (EGPRS) network. This QoS scheme is composed by an Admission Control (AC) and a Channel Allocation (CA) functionality, which are taking part in the connection establishment phase, as well as a QoS aware Packet Scheduler (PS) acting along the connection duration. The scheme is rounded off by the use of a Quality Control (QC) function, in charge of checking that the provided QoS by the system is in line with the negotiated QoS with the mobile station (MS). The performance of the proposed QoS framework has been evaluated by means of dynamic simulations. I. INTRODUCTION Multimedia streaming services are receiving considerable interest in the mobile network business []. The support of reliable real time services is a decisive aspect for the increasing migration towards packet based mobile networks. This kind of services is also technically applicable over evolving second generation (2G) and third generation (3G) wireless networks, thus streaming clients will soon be deployed in advanced wireless communication devices. Inside this new group of services, a variety of applications (e.g. audio and video on demand) with different traffic source statistical characteristics exists [2]. Therefore, in case of audio, the generated traffic is rather non-bursty whereas video traffic has a more bursty nature. Anyway, packet switched (PSW) bearers give more trunking gain and better resources utilization while circuit switched (CSW) bearers offer better performance for those services with stringent delay requirements. All the multimedia services are mainly characterized by the necessity from the network point of view to guarantee certain QoS requirements. The rise of these new multimedia services with stringent QoS requirements is converting the existing GSM/EDGE (Enhanced Data rates for GSM Evolution) networks into real 3G networks. This conversion implies the inclusion of several improvements in those networks, as the enhancement of the radio interface. One of those improvements is the necessity of a complete RRM framework aware of the QoS requirements imposed by the aforementioned new services. In this paper an Enhanced QoS (EQoS) framework for the transmission of streaming services over the EGPRS network is presented. The scheme consists of several improvements over existing RRM schemes with the aim of making them aware of QoS. That way, it is possible to support services requiring guaranteed data rates, as multimedia streaming. The next section shows the main characteristics of the streaming traffic class. In the third section the architecture of the mobile network is presented. The fourth section depicts in detail the EQoS concept. In the fifth section the simulation environment is presented, followed by the simulation results in the sixth section. The final section sum up the main conclusions obtained from these studies. II. STREAMING TRAFFIC CLASS The streaming traffic class (as defined in [3]) can be used for transporting real time video (or audio) information at human destination. The main characteristic of this traffic class is that the time relation between consecutive information packets (jitter) of the stream should be preserved. This means, the jitter introduced by the network to the end to end flow should be limited to preserve the real-time streaming characteristic of the stream. But as the stream is normally time aligned at the receiver (because of jitter compensating buffers at application layers), the acceptable jitter over the transmission media is higher than the one tolerated in other conversational applications, as e.g. videoconference. Ensuring low transfer delay of the stream packets in this traffic class is not as crucial as for conversational applications, since there is no interactive feedback from the receiver to the transmitter required. Main characteristics of the streaming traffic class are expressed through the QoS profile. An example is presented in Table, as well as some values as reference. The main idea for handling streaming traffic is that, because of the jitter compensating buffers in the system, once the bandwidth resources are reserved, the delay requirements are going to be fulfilled. Hence, the key point is how to guarantee the throughput parameter: guaranteed bitrate. Table. Example of Traffic Class QoS requirements QoS99 Parameter name Parameter value Traffic Class Maximum bitrate for uplink kbps Maximum bitrate for downlink 2, 4, 72 kbps Maximum SDU size 6 bytes Delivery of erroneous SDUs No SDU error ratio -2 Transfer Delay 2s Guaranteed bitrate for uplink kbps Guaranteed bitrate for downlink 6, 32, 64 kbps

2 The purpose of the guaranteed bitrate QoS parameter is indicating to the mobile network the minimum bitrate that the network has to guarantee. That means, when the traffic source is sending packets at an incoming rate up to the GBR, such bitrate will be ensure; however, when the incoming traffic rate exceeds the GBR, the network has only to assure the transfer of the GBR. In addition, the QoS requirements expressed by delay and reliability attributes only apply to the incoming traffic up to the GBR. As it is proposed here, guaranteed bitrate can be used to facilitate an AC based on available resources, and for resource allocation within the EGPRS network. The PS is taking care of guaranteeing the negotiated bitrate to all the admitted connections, while the QC function is guarding that the network is providing up to the negotiated GBR. III. MOBILE NETWORK A. Architecture of the system Fig. shows a general overview of the network architecture (described in [4]) involved in the transmission of streaming services over an EGPRS network. As shown, the streaming servers are connected to the Packet Core Network (CN) through the Gateway GPRS Support Node (GGSN), which acts as a logical interface with external elements, like external packet data network (e.g. the Internet). The Home Location Register (HLR) stores subscribed QoS profiles of the MS and is also part of the CN. The CN is connected to the GSM/EDGE Radio Access Network (GERAN) through the Serving GPRS Support Nodes (SGSNs), which act as "routers". The RAN is completed by the Base-Station Systems (BSSs), connected to the SGSN. BSSs are directly connected to the MSs over the radio link. When the user wants to initiate a multimedia streaming session, the MS opens a Packet Data Protocol (PDP) context and sends the application QoS requirements. When the SGSN receives such information, after authorizing the request by checking the subscribed QoS profile stored in the HLR, the SGSN sends the QoS profile to the BSS through the BSS Packet Flow Context (PFC) mechanism. This mechanism gives the BSS visibility of the QoS parameters, allowing the management of the traffic in a more accurate way according to the agreed QoS. B. EGPRS Network EGPRS provides an evolution in the GSM radio interface to allow the support of high-speed data services using the same TDMA multiplexing scheme over 2 khz carriers as used in GSM [5]. EGPRS facilitates the provision of data transfer by means of significant changes to both the core network and the radio access network. Enhancements to the RAN include the introduction of a new air interface, referred to as EDGE. This interface makes use of 8-PSK & GMSK modulation schemes. The user data rates for these schemes under error free transmission conditions vary from 8.8 kbit/s for MCS- to 59.2 kbit/s for MCS-9. The throughput available to the end user is further increase by multisloting, which allow several timeslots (TSLs) to be allocated to one user. MS U m BSS G b SGSN G x HLR Packet Core Network G i GGSN Server Fig.. System Architecture EGPRS uses a single protocol stack to support different services in the RAN. GERAN User Plane protocol stack consists of three layers: Packet Data Convergence Protocol (PDCP) layer, which maps network level characteristics onto radio characteristics; Radio Link Control (RLC) layer, in charge of connection control, segmentation, error correction and flow control; and Medium Access Control (MAC) layer, which provides data transfer, allocation of radio resources and measurement reports. RLC layer has three operation modes: transparent, unacknowledged, and acknowledged. In the transparent mode, RLC processes the data without adding any overhead. In the unacknowledged mode, RLC adds/removes a header to each RLC block, allowing to share the same physical resource for different RLC flows. In this mode, RLC layer retransmissions are disabled. In the acknowledged mode, RLC retransmissions are enabled and in-sequence delivery is guaranteed. MAC layer has two operating modes: dedicated and shared. In the dedicated mode, only one user can be assigned to a physical channel. In the shared mode, several users can be time multiplexed onto the same channel and an appropriate MAC scheduling algorithm shall be used to handle resource sharing and priority. IV. ENHANCED QUALITY OF SERVICE The Enhanced Quality of Service framework is a complete RRM scheme oriented for the transmission of GBR services over EGPRS. Different functionalities are involved in this process: an Admission Control/Channel Allocation scheme, in order to accept/reject new allocation requests; a Packet Scheduler in charge of providing to each ongoing connection the needed transmissions for guaranteeing the bitrate; and a Quality Control functionality, whose goal is to check that the provided QoS is in line with the subscribed QoS. EQoS will make use of acknowledged RLC mode of operation, due to the non-stringent delay requirement for streaming services. Moreover, shared MAC mode is used, allowing the multiplexing of several streaming users over the same channel. The use of shared channels leads to a better trunking efficiency, when comparing with dedicated channels, due to the bursty nature of the incoming traffic. A. Admission Control and Channel Allocation Admission control is the functionality in charge of estimating whether the network has enough available resources to allocate a new connection. Meanwhile, the channel allocation has the task of determining which are the

3 channels that have to be reserved for this new connection. In this QoS framework, both functionalities are included in the same scheme. AC/CA have the goal of ensuring the guaranteed bitrate requirements of the incoming users in the establishment phase, as well as estimating that those requirements are going to be fulfilled according to planning radio conditions meanwhile the connection is established. The decision of allocating or rejecting a new connection is based on three parameters: the needed capacity by the new connection, the used capacity by the guaranteed bitrate connections allocated in the same cell, and a safety margin. A new connection will be allocated in the system whether the needed capacity plus the used capacity and the safety margin is lower than the available capacity in the channel, being rejected in other case, as depicted in fig. 2. The needed capacity by the incoming user is defined as the percentage of channel capacity the new connection needs in order to fulfil its bitrate requirements. The calculation of this capacity is based on three different parameters: guaranteed bitrate requirements of the new user, multislot MS capacity and an estimation of the throughput the cell will provide to the new connection. The network planning is done in a way that certain bitrate (i.e. throughput estimation) is achieved by a determined percentage of the users (e.g. 95%) over a certain cell area. Hence, it is estimated at connection set up the capacity required to provide the guaranteed bitrate per TSL. The used capacity is defined as the addition of the percentage of channel capacity used by other GBR connections allocated in the same cell, and a safety margin (defined as a percentage of the estimated total capacity) to prevent in advance the possible variations in the channel capacity that each connection is currently seeing. The percentage of channel capacity used by each connection is calculated based on the GBR requirement, on the number of timeslots assigned to that connection, and on the channel throughput that is experiencing. This throughput is based on the information sent by the MS to the network by means of the PACKET DOWNLINK ACK/NACK message [6]. This message is used to indicate the status of downlink RLC data blocks received and to report the channel quality of the downlink. From this information, the network obtains the block error rate (BLER), which jointly with the modulation and coding scheme (MCS) that the connection has used during the last reporting period is used by the BSS to calculate the effective throughput the channels in which the connection is assigned have performed. The way of calculating this effective throughput is shown in (). [ BLER] ( ) EffectiveThroughput( kbps) = MCS( kbps) The effective throughput is stored in a sliding averaging window, which means not only the last reported throughput is utilized, but a certain number of consecutive measurements are used. It allows considering in the final measure the possible short-term radio link changes. B. Packet Scheduler The Packet Scheduler task is to govern the order and duration of the transmissions among the different ongoing connections allocated in one cell. As commented, several streaming connections can be allocated at the same time in the same timeslots, and, consequently, some control mechanism over the transmissions is necessary. Used Capacity Needed Capacity Safety Margin Fig.2. Needed and used capacity concepts in AC/CA For streaming connections, the role of PS is to provide the needed amount of transmission turns to those connections in order to fulfil their guaranteed bitrate requirements. The PS also delivers the remaining capacity among others non-real time connections and streaming connections whose incoming bitrate exceeds the guaranteed bitrate. The concept presented in this paper is based on the Deficit Round Robin (DRR) algorithm used in wireline networks [7]. Wireless channels (e.g. EGPRS channels) characteristics make necessary to modify the DRR algorithm. The main characteristic of an EGPRS channel is the fast variation of the capacity, due to bit errors, multislot capacity and modification of coding scheme. In addition, there are other aspects to be considered, like the long pauses in data transfer during handovers, or the necessity of preempting signaling transmission over data transmission. The main goal of this algorithm is to provide the guaranteed bitrate to all the traffic flows. This is achieved by means of bit buckets (see fig. 3). These buckets track the different data flows, and serve those flows that need transfer the capacity most urgently, in an order defined according to the scheduler priorities of the different traffic classes. Once all the throughput guarantees are fulfilled, the remaining capacity (the capacity not used for guaranteeing throughput) is distributed among the different allocated flows (up to the maximum bitrate requirement). LLC Queue # LLC Queue #2 LLC Queue #3 Quantum Zero Level Deficit Level Fig.3. Modified DRR Algorithm Selector Total Channel Capacity Selector considers queues with deficit level >

4 C. Quality Control function Along the connection, the QC function makes sure that the QoS provision in terms of throughput, delay and loss ratio is in line with the negotiated QoS requirements. When QC notices a degradation e.g. in the provided throughput, a process to maintain the negotiated QoS requirements starts. Therefore, the main task of the QC function is to check whether the negotiated QoS is being provided for each streaming connection. This function requires understanding about retransmissions to measure the real throughput of the connection. QC is responsible for reporting about degradation on the provided QoS with the aim of triggering proper actions to counteract such situation, as channel reallocation, cell reselection or QoS renegotiation. After monitoring, QC analyzes the statistics of the monitored throughput. When the incoming bitrate is larger than the GBR, the GBR is the target bitrate for the network. But when the incoming bitrate is lower than the GBR, the incoming bitrate will be the target bitrate that the monitored throughput must reach at least (i.e. target throughput = min{incoming, guaranteed bitrate}). In case the monitored throughput does not achieve the target throughput, QC considers there is a period of degradation. In case the degradation lasts more than a determined duration set as threshold, actions to counteract such situation are triggered. V. SIMULATION ENVIRONMENT The performance of the EQoS framework has been evaluated by means of dynamic system level simulations. All the simulations have been made over a macro-cellular scenario, which consists of 75 cells in a regular three-sectored hexagonal layout. Users are uniformly distributed through the system coverage area. The simulations have been run with 3 km/h user mobility [8]. One traffic TRX per cell has been used in the simulations, with a frequency reuse 3/9. The GSM air interface is modeled at burst level, including multipath fading, shadowing, distance attenuation, and both cochannel and adjacent channel interference. RLC/MAC protocols are implemented according to the specifications, including selective ARQ, retransmission bitmaps, polling and both transmitter and receiver sliding windows. Higher levels are partially modeled, including features as flow control mechanism at SGSN. In addition, acknowledged RLC mode and shared MAC mode are used for data transmission. Multislot operation is assumed for streaming services, meanwhile single slot operation is used for non-real time services. The simulations are focused on the downlink performance. The streaming traffic source is modeled with an autoregressive model for the packet size distribution with random variable following a Gamma distribution [9], and a deterministic packets interarrival time. The video codec is generating traffic at 6, 32 and 64 kbps as mean bitrate. Nonreal time traffic is modeled trying to use all the remaining capacity not used by streaming traffic. AC/CA is set in order to guarantee the mean streaming source bitrate, and a safety margin of 5% is used. PS is also set to guarantee the mean streaming bitrate. For non-real time traffic, a minimum throughput of kbit/s is assured. In order to measure the performance of EQoS, a satisfied user criterion has been defined: an streaming user is defined as satisfied whether the connection bitrate (total number of transmitted bits divided by the total service time) is above the 9% of the guaranteed bitrate. It is considered as an acceptable system if the 95% of the streaming users are satisfied, and the blocking level is below the 2%. VI. SIMULATION RESULTS Three different scenarios have been simulated, depending on the GBR requirement of the incoming streaming traffic: 6, 32 and 64 kbps. The idea is to show the behavior of the EQoS framework when varying the guaranteed bitrate. Fig. 4 shows the distribution of connection bitrate for the three different cases, meanwhile fig. 5 presents the corresponding capacity that EGPRS network is able to support for each guaranteed bitrate, always under a situation of acceptable system (95% of satisfied streaming users and blocking level below 2%). The EQoS RRM is able to guarantee the required bitrate in different situations. However the differences among situations can be found in the system capacity for different guaranteed bitrate services (fig. 5). Hence, the system is able to manage more traffic with lower bitrate (e.g. up to 8 users per cell at 6 kbps), because the statistical multiplexing gain is higher for such connections. PCDF Connection rate [kbps] 6 kbps 32 kbps 64 kbps Figure 4. Connection rate Distribution for different Guaranteed rate Cases Capacity [Users/Cell] kbps 32 kbps 64 kbps Figure 5. Traffic Capacity for different Guaranteed rate Cases

5 The corresponding non-real time traffic transmitted through the EGPRS network over the remaining capacity (i.e. left by streaming connections) is presented in fig. 6. As shown, the higher the streaming capacity, the lower the nonreal time capacity. In this figure, all the non-real time users are experiencing a connection bitrate at least above the minimum throughput, which is set to kbps and is guaranteed by the packet scheduler. Although streaming traffic does not have stringent delay requirements, in Fig. 7 the delay distribution of IP packets at link level is shown, in order to check how those delays are also fulfilled. This figure also provides some help for the dimensioning of the compensating buffers in (E)GPRS mobile networks. It is also observed in fig.7 that the experienced delay in case of connections at higher GBR is lower. Attending to the 95 th percentile of the delay distribution, in case of 6 kbps is around 7 ms, whereas in case of 64 kbps is around 35 ms. The reason is the different amount of connections in the system, a higher number of low bitrate connections are multiplexed in the same timeslot, when comparing with high bitrate connections. AC can control this delay by restricting the admission of streaming connections. Modifying the Link Adaptation algorithm, determining the most appropriate thresholds to select the transmission coding scheme can also reduce the transmission delay. Spectral Efficiency [kbit/s/cell/mhz] kbps 32 kbps 64 kbps non-real Time Figure 6. and non-real time Spectral Efficiency for different Guaranteed rate Cases PCDF Frame Delay [sec] 6 kbps 32 kbps 64 kbps Figure 7. Frame Transfer Delay Distribution for different Guaranteed rate Cases VII. CONCLUSIONS A complete RRM scheme oriented for the transmission of guaranteed bitrate services over EGPRS has been presented in this paper. The Enhanced Quality of Service framework consists of different functionalities: an Admission Control/Channel Allocation scheme, in order to accept/reject the new allocation requests; a Packet Scheduler in charge of providing to each ongoing connection the needed transmissions for guaranteeing the bitrate as well as delivering the remaining capacity among non-real time connections; and a Quality Control functionality checking that the provided QoS is in line with the negotiated QoS requirements. Simulation results show the correct performance of the EQoS scheme, both guaranteeing certain bandwidth and a delay lower than the requirements. The use of shared channels and the multiplexing of streaming connections lead to a better trunking efficiency, when comparing with dedicated channels. ACKNOWLEDGMENT This work has been performed as part of the cooperation agreement between Nokia and the University of Malaga. This agreement is partially supported by the Program to promote technical research (Programa de Fomento de la Investigación Técnica, PROFIT) of the Spanish Ministry of Science and Technology. The authors would like to thank Javier Romero and Julia Martínez for their support. REFERENCES [] M. Margaritidis and G. C. Polyzos, "MobiWeb: Enabling Adaptive Continuous Media Applications over 3G Wireless Link," IEEE Pers. Commun., vol. 7, Dec. 2, pp [2] A. Mena and J. Heidemann, "An Empirical Study of Real Audio Traffic," IEEE INFOCOM 2, vol., pp. - [3] 3GPP TS 23.7 v5.. (2-6), "QoS Concept and Architecture (Release 5)" [4] 3GPP TS 23.6 v3.2., 2, "General Packet Radio Service (GPRS); service description; stage 2" [5] T. Halonen, J. Romero, J. Melero, GSM, GPRS and EDGE Performance, Evolution Towards 3G/UMTS, Wiley, 22 [6] 3GPP TS 4.6 v8.. (2-9), "General Packet Radio Service (GPRS); Mobile Station (MS) - Base Station System (BSS) interface; Radio Link Control/Medium Access Control (RLC/MAC) protocol (Release 99)" [7] M. Shreedhar and G. Varghese. "Efficient Fair Queuing Using Deficit Round-Robin". IEEE Transactions on Networking, Vol.4 No [8] Universal Mobile Telecommunications System (UMTS): Selection procedures for the choice of radio transmission technologies of the UMTS. ETSI Technical Report 2 V3.2. (998-4) [9] S. Xu and Z. Huang. "A Gamma Autoregressive Video Model on ATM Networks". IEEE Transactions on Circuits and Systems for Video Technology, Vol.8,Issue 2, April 998, pp