Monday, May 28, 2012

Unsolicited Grant Service with Activity Detection

The UGS-AD algorithm is designed to support real-time service flows that generate fixed-size data packets on a semi-periodic basis (e.g., VoIP using on–off voice codec). It incorporates activity detection, which makes it suitable for use with on/off voice codecs. The algorithm uses combined features of UGS and rtPS. UGS-AD has two scheduling modes: UGS and rtPS, and can switch between these modes depending on the status of the voice users (on or off). On initialization of VoIP services, this algorithm starts with the rtPS mode. While in rtPS mode, if the voice user requests bandwidth size of zero bytes, the BS maintains this (rtPS) mode. However if the user requests bandwidth size greater than zero, the BS switches its mode to UGS. While in UGS mode, if the voice user requests bandwidth size = 0, the BS switches to rtPS, and if the user requestsbandwidth greater than zero, the BS stays in UGS. By switching between rtPS and UGS modes, the UGS-AD algorithm significantly addresses the problem of UL resources wastage in the UGS algorithm, and the MAC over head and access delay in the rtPS. This is however only for the case where the voice user uses voice codecs with only two data rates (on–off). Where voice codecs with variable data rates like EVRC is used, resources wastage still occur in the UGS-AD algorithm. In this case, the wastage occurs during the on duration, when full resources is assigned eventhough the variable data rate of voice codecs means that it will not operate at full rate for all of the time the resources is allocated. The operation of the UGS-AD algorithm is illustrated in Figure 1.

Figure 1: UL resource allocation using UGS-AD algorithm.

Thursday, May 24, 2012

Real-Time Polling Service | QoS Scheduling

The rtPS algorithm is designed to support real-time service flows, such as MPEG video or tele-conference, that generate variable size data packets periodically. In this algorithm, the BS assigns UL resources that are sufficient for unicast bandwidth request to the voice users. This is called the polling process. The duration for which the BS continues to poll an SS with rtPS connection is negotiated in the initialization process of the connection. The SSs utilize the assigned polling resources to send their bandwidth requests, reporting the exact bandwidth need for their rtPS connection. The BS in response then allocates the exact bandwidth requested to the SS for transmission of the data. Figure 1 illustrates this dynamic polling process.

Figure 1: Polling process in rtPS.
Because rtPS always carry out polling process, it is able to adaptively determine suitable resource allocation from frame to frame. This adaptive request-grant process goes on until the connection is terminated. Because of the dynamic request-grant process, the algorithm has more optimum data transport efficiency than the UGS algorithm. The algorithm is able to dynamically follow all data rate of the voice codec without any resource wastage as illustrated in Figure 2 (allocated and utilized resources are equal). This is a major advantage over the UGS algorithm. The drawback of the rtPS algorithm however is that the dynamic polling process causes MAC overhead and access delay. Hence rtPS has more MAC overhead and larger access delay than the UGS.

Figure 2: UL resource allocation using rtPS algorithm.

Sunday, May 20, 2012

Unsolicited Grant Service

The UGS algorithm is designed to support real-time service flows, such as Voice over Internet Protocol (VoIP), that generate fixed size data packets periodically. BS periodically assigns fixed-size grants to voice users. These grants are sufficient to send voice data packets generated by the maximum data rate of enhanced variable rate “voice” codec (EVRC). The grant period are negotiated during the initialization process of the connection. Thus, MAC overhead and UL access delay caused by bandwidth request process are minimized. The drawback of the UGS algorithm is the following. Generally, voice users do not always have voice data packets to send throughout the duration of a connection, because voice users have frequent silence periods. A typical voice codec switches intermittently between “on” and “off” states as illustrated in Figure 1. While in “on” state, popular voice codecs like the EVRC also have variable data rates. For example, the EVRC operates at 1/8 of the full data rate during the off state, while the device has three different rates during the on state (rates 1, 1/2, and 1/4). Therefore for a UGS algorithm that reserves a flat amount of resources capable of sending data at the maximum rate of EVRC periodically, a significant amount of UL resources is wasted when the codec is in silence (or off) mode as well as when the codec is on but not operating at the full rate. This is illustrated in Figure 2. A number of other algorithms have thus been designed to adaptively determine the actual UL needs of each connection during frame periods, so as to minimize these resources wastage.

Figure 1: Voice codec status.

Figure 2: UL resource allocation using UGS algorithm.

Wednesday, May 16, 2012

Scheduling Algorithms

In the WiMAX standard (802.16), four scheduling classes have been defined: Unsolicited grant service (UGS), real-time polling service (rtPS), nonreal-time polling service (nrtPS), and best effort (BE). As illustrated in Figure 1, each traffic connection is associated with one of the four scheduling services, and the SS scheduler selects packets to be transmitted from each queue based on the scheduling policy employed. Usually, the scheduler selects packets to be transmitted from the highest priority queue that is not empty. Transmission of packets from lower priority queues are postponed until there is no packet available to send from a higher priority queue. Since UL traffic is generated at SS, the SS scheduler is able to arrange the transmission based on the up-to-date information on the current numbers/status of UL connections, which help to improve QoS performance. In the following, we review the various scheduling algorithms provided in the standard for handling the transmissions of packets belonging to these various services.

Figure 1: UL scheduling at the SS.

Saturday, May 12, 2012


IEEE 802.16 supports fixed-length frame, with flexible (adaptive) DL/UL resource usage ratios. The BS adaptively adjusts DL and UL subframe lengths on a frame-by-frame basis depending on the DL/UL traffics and channel conditions. Typically, the DL:UL resources can be varied from 3:1 to 1:1 in a PMP WiMAX network. Figure 1 illustrates the fixed-length frame in the PMP WiMAX network, and the flexible DL/UL subframes. The figure also depicts the network entry process for subscriber stations (SS) and the scheduling periods for assigning transmission opportunities to SS already initiated into the network. For access (PMP) mode, new SS detects preamble and frame control header (FCH), and identifies the number of DL burst transmissions from the DL MAP in the FCH. At the end of the last DL burst (Figure 1), new SS uses a contention period to exchange network entry request signal with the BS. If successful, the BS process the request and sends entry instruction (assigned DL/UL transmission opportunities, power, etc.) in the DL/UL MAPS of the next frame, and the SS gets initiated into the network. For the mesh mode, new SS waits for network entry signal broadcast at the beginning of a frame, to which they can respond within a specified period. Scheduling process is used for initiating new SS into the network. SS transmits on the scheduled slots.

Figure 1: Adaptive DL/UL subframes in WiMAX standard.
In the WiMAX standard (802.16e), UL and DL assignments are based on time division multiple access (TDMA). In each frame, the BS scheduler assigns UL and DL transmission opportunities to SS until their negotiated data periods expire. The resources given to an SS for its data transmission are both in the frequency and time domain. WiMAX MAC thus supports frequency-time resource allocation in both DL and UL on a per-frame basis. The resource allocation is delivered in media access protocol (MAP) messages at the beginning of each frame. Therefore, the resource allocation can be dynamically changed frame-by-frame in response to traffic and channel conditions. Additionally, the amount of resource in each allocation can range from one slot to the entire frame in the time domain, and from one subchannel to the entire subchannels in an OFDM symbol, in frequency domain. Also WiMAX employs fast scheduling both in the DL and UL to respond to fast variations in channel conditions. This fast and fine granular resource allocation allows superior QoS for data traffic in a bursty traffic and rapidly changing channel condition. The fundamental premise of the IEEE 802.16 MAC architecture is QoS. It defines services flows which can map to Diffserve code points or MPLS flow labels that enable end-to-end IP-based QoS. Additionally, subchannelization and MAP-based signaling schemes provide a flexible mechanism for optimal scheduling of space, frequency, and time resources over the air interface on a frame-by-frame basis. This flexible scheduling allows QoS to be better enforced and enable support for guaranteed service levels including committed and peak information rates, latency, and jitter for various types of traffic on a customer-by-customer basis.

Wednesday, May 9, 2012


FEC allows the WiMAX MAC layer to detect errors introduced during the transmissions of frames over the air link. There are three methods of FEC specified in the WiMAX system; Reed-Solomon concatenated with convolutional code (RS-CC), block turbo code (BTC), and convolutional turbo code (CTC). RS-CC is mandatory, while BTC and CTC are made optional due to their complexity, eventhough they provide 2–3 dB better coding gain than RS-CC. For the 802.16e, a hybrid ARQ (H-ARQ) has been included as an optional feature. There are three types of H-ARQ, classified based on the manner in which they handle the retransmissions. 

Type I H-ARQ retransmits lost or unacknowledged blocks using chase combining in which the old erroneous block is stored at the receiver and compared with the retransmitted copy. This helps to increase the probability of successful decoding at the FEC block during the retransmission attempts. Type II/III H-ARQ uses incremental coding rate to ensure successful decoding at the FEC block during the retransmission attempts. Rate adaptation works hand-in-hand with the FEC block in the WiMAX system. When a user experiences good channel condition, it is desirable to exploit these peaks in the channel gain to increase throughput. 

This is achieved by having the SS increase the coding rate, e.g., from rate 1/2 code to rate 1/4 code, so that more information bits can be transmitted per channel use while still keeping to the target bit error rate (BER). When the channel degrades, the rate is reduced back to the next minimum to ensure that the target BER is met. This dynamic process is carried out on a frame-by-frame basis in the WiMAX system, using the flexibility provided by MAP signaling to adaptively adjust the UL/DL rates.

Sunday, May 6, 2012


In wired networks the channel impairments tend to be constant or at least very slowly varying. Wireless networks in contrast are well known for rapidly fluctuating channel conditions even when the transmitter and receiver are stationary. Broadly speaking, the lower the modulation and coding rate, and the higher the transmitted power, the more channel fading a system can tolerate and still maintain a link at a constant error level. It is desirable therefore to be able to dynamically change the transmitted power, modulation, and data rate to best match the channel conditions at the moment to continually support the highest capacity channel possible. WiMAX systems support adaptive modulation and coding on both the downlink (DL) and uplink (UL) and adaptive power control on the UL. Adaptive modulation allows the WiMAX MAC layer to adjust the signal modulation rate depending on the channel or radio link quality. When the channel quality is good, the MAC layer chooses the highest modulation rate, e.g., 64QAM, giving the system the highest throughput. When the channel quality degrades, the MAC layer reduces the modulation rate, e.g., 16QAM, reducing the throughput. In practice, adaptive modulation and coding rate control are used in conjunction with power control. In the PMP network deployments with multiple users in a cell serviced by a BS, when a link degradation arises for a user, the BS first increases the transmitted power of the user to provide extra link budget gain, until it reaches the maximum permitted. If the received signal quality does not improve, then the coding rate is reduced. Extra redundancy is added to provide more coding gain for better error correction performance. If the received signal quality still does not improve, then the modulation rate is reduced as a last resort (as this significantly affects the throughput than others). Similar (reverse) process is also followed when link quality appreciates. For WiMAX mesh networks using the amplify-and-forward relaying option, mesh relaying cannot exploit adaptive modulation technology because relaying nodes are not able to decode the contents of the received OFDM symbols to retrieve the modulated data and remodulate them at higher or lower rate, to increase or reduce the transmission rate (or throughput) of the mesh streams in response to link quality condition. However for mesh networks using the decode-and-forward relay option, adaptive modulation and coding rate control can benefit the mesh relaying operation as mesh nodes can decode the mesh data streams and adjust the coding and modulation rate, depending on the forwarding link quality. For example in Figure 1 a relay node decodes a data stream originally transmitted from the source node using 16QAM modulation, and remodulates the data stream using 64QAM as it has good channel quality to the destination node that can support this modulation rate. This results in fast and efficient use of the mesh links. Power control is applicable in WiMAX networks (PMP and mesh) in two ways: One, when nodes are transmitting data, they are regulated to transmit only the minimum power required to achieve successful reception at the receiver. Two, when mobile nodes do not have data (mesh relay or access service data) to transmit or receive, they go on sleep modes to save battery life.

Figure 1: Adaptive modulation at WiMAX mesh nodes.
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