A RLC/MAC PROTOCOL ARCHITECTURE FOR A WIRELESS IP NETWORK

Transcription

1 A RLC/MAC PROTOCOL ARCHITECTURE FOR A WIRELESS IP NETWORK Jamil Y. Khan School of Electrical Engineering & Computer Science, University of Newcastle, Callaghan, NSW 238, Australia, jkhan@ec .newcastle.edu.au Abstract - This paper proposes a RLC/MAC (Radio Link Control/Multiple Access Control) protocol architecture suitable for a wireless IP network. The RLC and MAC protocols have been designed to support packet transmission on an air interface. The MAC protocol uses a reservation technique to increase the multiplexing efficiency by minimising collisions. The RLC protocol is based on selective repeat technique; optimises its efficiency by using the efficient architecture of the MAC protocol. Simulation results are presented to evaluate the performance of the combined protocol. Keywords RLC, MAC, Wireless, IP, Air interface. I. INTRODUCTION Success of second generation mobile telephone systems, phenomenal growth of mobile and internet users and flexibility of IP based multimedia applications are the major issues driving the growth of next generation wireless networks []. Significant number of third generation and most likely all fourth generation wireless networks will use packet switched communication architecture to transmit multimedia information. Gradual shift towards All IP (Internet Protocol) communication architecture will require advanced packet based air interfaces capable of supporting end-to-end packet communication. Transporting IP packets on an air interface could be an option to support end-to-end packet based services. However, there are several reasons why the above option is not always suitable. Two of the main reasons are the reduced spectral efficiency due to large IP headers and high packet error rate due to transmission channel conditions. Packet header in an IP based multimedia network contains either TCP/IP headers for non-realtime traffic or UDP/RTP/IP headers for realtime services such as voice over IP (VoIP). The size of the combined UDP/RTP/IP headers is at least 4 bytes for IPv4 and at least 6 bytes for IPv6, while voice payload will be very small compared to this header size. For example a ms speech block coded with an 8kbs speech coder and a half rate convolutional coder will generate a payload of 2 bytes. Hence transmitting a packet with full IP header will greatly diminish the spectrum efficiency of any air interface. To reduce the size of an IP packet header, it is possible to use header compression and decompression techniques [2]. Header sizes can be reduced by removing redundancy in the original header and/or removing field information and thereby losing functionality. The header compression algorithms maintain a context, which is essentially the uncompressed version of the last transmitted header at each end of the transmission channel over which compression occurs. The compressed header solely carries information about changes to the context. In case of a lost compressed header the context on the downstream cannot be updated properly. The IETF (Internet Engineering Task Force) proposed compression algorithm CRTP (Compressed RTP), which compresses UDP/RTP/IP header from 4 bytes to 2 bytes, is vulnerable to transmission channel errors. Alternate compression scheme known as ROCCO (RObust Checksum-based header Compression) has been developed for cellular mobile telephony usage, which shows a lower packet loss due to channel error [2]. However at -3 BER (Bit Error Rate) about.7% - % of packets could be lost when the ROCCO compression technique is used [2], [3]. A packet based system could inherently lose packets during access phase as well as due to blocking [4]. Any additional packet loss due to the header compression can further reduce the Quality of Service (QoS). On the other hand for TCP/IP data packet transmission it is possible to transmit longer packets using a high payload to header ratio. However long packets are subject to longer retransmission delay on a wireless channel [5]. To avoid above problems it is possible to transmit users information packets on an air interface using IP compatible packets to the base station (BS). The base station can generate an IP packet by encapsulating several MAC packets with the appropriate UDP/RTP/IP or TCP/IP headers before transmitting IP packets on the radio access network (RAN). Similarly header could be stripped off and an IP packet can be defragmented at the BS before transmitting MAC packets on the air interface to the mobile users on the downlink. If the encapsulated IP packet is destined for a fixed network terminal then the complete IP packet can be delivered to the addressed terminal. The structure of the paper is as follows. Section II describes the RLC/MAC protocol structure used for IP packet transmission. Section III describes the IP packet transmission technique using the proposed RLC/MAC protocol architecture. Section IV presents simulation results and brief discussion is presented in section V. II. RLC/MAC PROTOCOL ARCHITECTURE This section describes the joint design of RLC/MAC protocol. The RLC protocol is based on a window ARQ /2/$7. 22 IEEE PIMRC 22

2 (Automatic Repeat request) technique and the MAC protocol is based on a block reservation technique known as the BRTDMA (Block Reservation Time Division Multiple Access) protocol. The main advantage of the combined protocol is minimum signalling requirements for the RLC layer and deterministic packet transmission delay for multiservice traffic while retaining high multiplexing efficiency. A. MAC Protocol The MAC protocol known as BRTDMA is based on the PRMA++/TDMA architecture use reservation and multipriority slot allocation technique to support different quality of service (QoS) requirements. The multiplexing architecture of the BRTDMA protocol is shown in figure. The R slot is used to transmit reservation requests by the mobile stations using the S-ALOHA protocol and I slots are used to transmit information in reserved modes. A slots on the downlink are used for signalling and I slots are used for downlink information packet transmission. FP and FP ak are the paired fast signalling channels. to transmit data traffic. The reservation length is kept constant irrespective of data burst length. Block reservation has the advantage of reduced contention and predictable data packet or data burst transmission delay. This reservation technique will be further discussed in the RLC protocol description. The multipriority slot allocation algorithm uses an allocation queue with two segments, which is kept at the base station. The allocation queue is divided into two segments; the head of the queue of length L packets is used as the slot allocation sub-queue and the rest of the queue is used as the storage as shown in figure 2. Successful traffic slot request packets are filed in the storage sub-queue as soon as a new request packet arrives. Requests are then sorted and transferred to the head of the queue for allocation. Incoming requests R R I I R I I FP ak UP Link I I A A I FP I DOWN Link Figure : BRTDMA protocol frame and slot structure The BRTDMA protocol has been developed to cater for multiservice traffic. It offers reservation to both realtime and non-realtime traffic. For speech traffic the reservation is offered for every talkspurt and for a non-realtime traffic reservation offered for a block of slots [6]. Base station allocates reserved information slot (I) based on priority and the priority is altered with the changing traffic situation. Multipriority slot allocation technique has been used to minimize packet losses from realtime sources [6]. The multipriority slot allocation technique replaces the FIFO (First In First Out) algorithm generally used at the Base Stations to allocate information slots. Above two features i.e. block reservation and multipriority slot allocation techniques of the MAC protocols are used to increase the multiplexing efficiency as well as the RLC protocol use these features to minimize signalling and retransmission delay. B. Block Reservation Technique The MAC protocol use block reservation for non-realtime sources such as data traffic sources [6]. A requesting data terminal obtains a number of reserved slots (one slot/frame) Storage segment Allocation segment Figure 2: Multipriority queue for allocating traffic slots. The sorting cycle is synchronized with the request arrival process. Request packets can only arrive at the base station using R slots on the UP link. The sorting algorithm updates the allocation segment of the queue after every transmission frame instead of every new request packet arrival. Traffic slots are allocated using the FIFO technique after the slot requests are placed in the allocation queue. During the sorting cycle both the entries in the storage sub-queue and the allocation sub-queue are sorted. In each sorting cycle slot requests are sorted according to the service priority P service, retransmission number RT, arrival time T arr and the reposition count RC. Slot request packets arrive at a base station with two attributes, those are service priority P service and the retransmission number RT. Also at the arrival of a request packet a BS notes the arrival time, which is used for queue sorting. The service priority is attached to a terminal by the call admission procedure. In case of initial access a terminal uses a default value of P service, which is based on the type of service the terminal requires from a network. The value of RC(station) is used by the sorting algorithm to reposition slot requests in the queue. Initial value of RC(init) is a system parameter that is related to the range of services. The value of RC(station) is decremented by the sorting algorithm. When the RC(station) value reaches its minimum the slot allocation priority becomes highest. The value of RC is decremented using the equation. K is the sensitivity factor that is used to control the number of time a slot request could be repositioned.

3 RC( station) = RC( init) Pservice * K RT () The value of K is controlled by a timer, which calculates the length of time a request packet has waited in the allocation and/or storage queue. Value of K is could be changed to promote the position of a request in the allocation subqueue. The value of K could also be biased based on the service types. To sort slot request packets; BS place request packets based on the value of RC. Slot requests may change their groups and positions after every sorting cycle. C. RLC Protocol The RLC protocol used here is a window based selective repeat ARQ (Automatic Repeat request) technique. The RLC layer utilise the block reservation technique and the multipriority slot allocation technique to minimize signalling and retransmission delay. The ARQ algorithm uses a selective retransmission scheme using a fixed window size as shown in figure 3. A data terminal receives a reservation of M slots to transmit a data burst or a segment of a data burst in M consecutive transmission frames using one slot per frame. The RLC layer use receives an acknowledgment after transmitting the whole block of data i.e. M data packets. The acknowledgement (ACK) packet is sent by the BS using an A slot on the downlink. The ACK packet contains a number of flag bits to indicate whether a particular packet transmission was successful or unsuccessful. Unsuccessful packets are retransmitted by appending those packets with the next data block as shown in the figure 3. The acknowledgement packet format is discussed in section III. When retransmission packets are appended with a data burst then the service priority (P service ) of that particular station is modified to reduce the slot allocation delay. UP LINK Retransmission packet ACK Packet DOWN LINK Figure 3: Block reservation technique. A retransmission timer is used to compensate a lost ACK packet. Retransmission timer value is selected based on the expected ACK packet arrival time that is given in equation 2. Parameter T L is the tolerance limit. Due to congestion on the down link signalling channel it may not be possible to transmit an ACK packet in time. In highly loaded situation a terminal can use longer T L value. Should an ACK packet is lost; a transmitting terminal will initiate retransmission of all packets in that reserved block after the expiry of the timer. acktimer ( f + t p ) + TL t f T = M t * (2) where t f represent frame length and t p propagation delay respectively. III. IP PACKET TRANSMISSION As discussed before that transmitting an IP packet over an air interface could reduce spectral efficiency as well as quality of service (QoS) due to channel errors. Instead of transmitting an IP packet on the air interface IP packets could be generated at a BS using an IP compatible air interface. An IP packet will be generated at the BS by accumulating M number of MAC packets received via the air interface. In this case the IP packet encapsulation time will be determined by the transmission frame length, propagation delay and number of MAC packets used to create an IP packet. In this work it is assumed that an IP packet will be generated after receiving M packets (length of the block reservation). To facilitate the IP packet encapsulation at the BS a mobile terminal will send following information during the connection request using a R slot on the UP link. Information transmitted on the UP link during connection request: Terminal/ Source ID and Destination ID. This information could be directly inserted in the IP header. Connection type and QoS information. This information could be translated into Type of Service and Time to Live fields in an IPv4 header or into Traffic Class and Flow label information in IPv6 header. Payload Length/Data Segment Size. This information will be used by the BS to calculate the expected number of IP packets from different active sources and may be used for resource allocation activities. Payload status. With this information MS will notify the BS whether the current data block contain any retransmitted packet(s) or not. If a retransmitted packet(s) is included within this burst then the BS will modify the P service value for shorter access time. On receiving a connection request packet, a BS will file the request in the slot allocation queue and allocate an available traffic slot using the A slot on the downlink. When a slot is allocated to a user the BS will generate a logical connection ID that is used by the mobile users to transmit MAC packets to the BS. At the BS the logical connection number is used to map the MAC packets to an appropriate

4 IP packet at a BS. On the downlink the BS transmit following information when a slot is allocated. Reserved slot number. Reservation length i.e. how many slots will be available to transmit the current burst. In case of a speech terminal it will be open i.e. the reservation will be relinquished by the MS and for data transmission the BS will indicate the length. A logical connection ID, which will be included in the header of the MAC packets. New P service value in response to the payload status information. The BS is also responsible for transmitting an ACK packet for data terminals transmitting data packets using a block reservation. An ACK packet is transmitted on completion of block transmission containing following information. Logical connection number. Logical connection number can be used for subsequent reservations. Status of each transmitted packets in a block indicated by a separate status bit. IV. SIMULATION RESULTS Performances of the MAC and the RLC protocols were evaluated using a simulation model. Figure 4 shows the speech packet loss pattern for different values of data block reservation length (M) and input traffic. In this simulation the number of speech terminal is kept constant to 45 and number of data terminal is varied. Data terminals were modelled using a negative exponential distribution with different interarrival time and variable burst sizes between Kbs to kbs. Results show that both longer (M=25) and shorter (M=) reservation block sizes introduce higher packet losses. Shorter block length introduces higher packet losses because of higher contention during the access period whereas a longer reservation length introduces higher blocking resulting speech packet losses. In this simulation speech terminals were given higher priority by using a shorter retransmission time, higher value of RT and P service. Simulation results were also obtained for variable data block sizes. In this case the reservation block length (M) is increased if the total traffic load is less than 7% of the system capacity. Load is sampled after every adaptation period. In this simulation adaptation period of 75 ms is used to adapt the value of M. Variable data block length utilises unused channel capacity to transmit larger data blocks thus reducing end-to-end data block transmission delay. For lower network traffic load, variable data block size allows formation of large IP packets at a BS without penalising speech terminals Next we obtain the data block transmission delay. IP packet encapsulation delay is consists of data block transmission delay and the corresponding packetisation delay. Figure 5 shows the normalised data block transmission delay for different values of M. Assuming negligible packetisation delay at the BS, IP packet encapsulation delay at the BS will be same as the data block transmission delay. Normalised block delay is derived by dividing the total block transmission delay by the corresponding data block size. Figure 5 shows that the shortest block size (M=) introduces higher delay particularly at higher load whereas a longer block size (M=25) introduces shorter delay. For multiservice traffic the value of M need to be optimised. Results also shows that variable data block size can offer optimum performance for both speech and data terminals. In this simulation packet length of 222 bits are used where 6 bits are used as a guard band to support cell size of km and 6 bits are used an overhead indicating the logical connection number and CRC bits. Each data packet carrying a payload of 2 bits, an IP packet of 32 bits or 4 bytes can be generated within 2 ms using a data block length of 6. At lower traffic load longer block size can be used without increasing any speech packet loss. % of speech packet loss Figure 4: Speech packet loss for different data block reservation length. Number of voice terminals is kept fixed to 45. Normalised block delay (ms) No. of data terminals 5 M=6 M=25 M= M=variable No. of data terminals M=6 M=25 M= M=variable Figure 5: Normalised data block transmission delay.

5 To evaluate the performance of the retransmission scheme of the RLC protocol, IP packet encapsulation delay was measured for frame error rates of % and 2%. In this case it is assumed that a single retransmission is required to successfully transmit a corrupted packet. Figure 6 shows the IP packet encapsulation delay for different frame error rate (FER). For M =6, FER of % means 2 packets require retransmission. This delay includes the original data block transmission delay, packet retransmission delay(s) and additional access delay. The normalised delay is derived by dividing the total delay by the corresponding block size. Different delay components of the IP packet encapsulation delay are shown in equation 3, where T bd is the original block transmission delay, T PR is retransmitted packet transmission delay, T access is subsequent data block access delay, k is the number of retransmitted packets and n is the number of retransmission required to successfully transmit all corrupted packets. Simulation results showed in case of a retransmission the access delay becomes the dominant component of the end-to-end delay at high traffic load. If a received data block contains several packets with error then the BS will wait until the corrupted packets are retransmitted and received. Retransmitted packets are included in a new block where retransmitted packets are placed in the head of the new block. T Normalised delay (ms) encap = T bd + k i= n ( T ) + ( T ) PR i j= access No. of Data Terminals 25, % 25, 2% 25, % 6, % 6, 2% 6, % j (3) RLC protocol requires minimum signaling. The proposed protocol architecture is an alternative solution to the end-toend IP packet transmission. Simulation results presented in the reference [5] shows that for better spectral efficiency and minimum end-to-end delay it is necessary to fragments large IP packets in to smaller fragments before transmission of the air interface. Large packets on a wireless link introduce higher probability of packet error wasting significant transmission capacity. The proposed technique can be used both in the TDMA and CDMA based systems. Also the proposed technique can easily be configured to support a number of different priority traffic rather than only two priorities presented in this paper. The proposed algorithm can be further improved by combining link adaptation procedure and hierarchical forward error correction techniques. REFERENCES [] M. Frodigh, et.al, Future Generation Wireless Networks, IEEE Personal Communications, vol:8, no:5, pp.-7, October 2. [2] K. Svanbro, J Wiorek and B Olin, Voice over IP over Wireless, Proceedings of the IEEE PIMRC2, London, UK, vol. I, pp , September 2. [3] A. Cellatoglu, S. Fabri, S. Worrall, A. Sadka and A. Kondoz, Robust Header Compression for Real-Time Services in Cellular Networks, Proceedings of 2nd Int.Conference on 3G Mobile Communication Technologies, London, UK, pp , March 2. [4] J. Dunlop, P. Cosimini, D. Maclean, D. Robertson and D. Aitken, A reservation based access mechanism for 3 rd Generation Cellular Systems, IEE Electronics & Communication Journal, vol:5, no:3, pp. 8-86, June 993. [5] A. P. Prasad, Optimisation of Hybrid ARQ for IP Packet Transmission, Wireless Personal Communications, vol:6, no:3, pp , March 2. [6] J. Y. Khan, Performance of a Multipriority Resource Allocation Technique for a Packet Switched Wireless Network, Proceedings of PIMRC2, London, UK, vol 2:, pp , September 2. Figure 6: IP packet encapsulation delay. V. DISCUSSIONS A simple RLC/MAC protocol is presented in this paper. The MAC protocol is based on reservation based packet transmission technique incorporating an advanced queuing technique that can alter priority to different types of traffic based on its QoS requirements. The combined protocol has two main advantages, those are a) Predictable transmission delay for all type of traffic sources (i.e. voice, data), b) the