Professor, Dept. of Computer Science and Engineering, Sri Siddhartha Institute of Technology, Tumkur,

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1 Computing For Nation Development, March 10 11, 2011 Bharati Vidyapeeth s Institute of Computer Applications and Management, New Delhi Techniques to Improve Performance of VoIP over e WLAN D. Ramesh 1, B.P.Mallikarjunaswamy 2 and B.R.Prakash 3 1 Research Scholar, Department of Computer Science and Engg., Kuvempu University, Shankargatta 2 Professor, Dept. of Computer Science and Engineering, Sri Siddhartha Institute of Technology, Tumkur, 3 Faculty, Dept. of Master of Computer Applications, Sri Siddhartha Institute of Technology, Tumkur, 1 rameshd_ssit@yahoo.com, 2 drbpmswamy@rediffmail.com and 3 brp.tmk@gmail.com ABSTRACT In today s Internet, the emerging widespread use of real-time voice, audio, and video applications makes Quality of service (QoS) a key problem. Meanwhile, the Internet is getting heterogeneous due to the explosive evolution of wireless networks. Quality of service (QoS) is an important consideration in networking, but it is also a significant challenge. Providing QoS guarantees becomes even more challenging when you add the complexities of wireless networks. This paper presented a MAC protocol that provides effective QoS for VoIP over WLAN. The characteristics of our proposed protocol are (1) No modification of access points, (2) No H/W modification of VoIP STAs and (3) Backward compatibility in order to minimize the costs of development and introduction. 1.INTRODUCTION The current Quality of Service (QoS) standard for wireless networks from the widely-used e family is IEEE e, The scope of this standard is to enhance the existing Media Access Control (MAC) so as to improve and manage QoS, to expand support for LAN applications with QoS requirements and provide classes of service. Wireless VoIP, typically over WLAN, is becoming increasingly popular, but even further elevates the challenges of delay and loss reduction. Degradation of speech quality caused by packet delay and loss of voice traffic is still one of critical technical barriers of the VoIP system. Furthermore, apart from these limitations WLANs will need to support a large number of concurrent VoIP communications since VoIP is spreading rapidly especially in public Spaces. These motivations led us to study the VoIP capacity in IEEE e WLAN and to investigate increasing this capacity by reducing VoIP codec rate while maintaining an overall good quality. 2.VOIP SYSTEM OVERVIEW This section introduces a short description of VoIP basic mechanisms and the E-Model used for objective assessment of audio quality. 2.2VOIP TRANSMISSION NETWORKS The commonly used VoIP codecs are G.711, G.729 and G The traditional sample-based VoIP encoder G.711 uses Pulse Code Modulation (PCM) to generate 8 bits samples per ms, leading to a data rate of 64 kb/s. Recent framebased encoders can be used in order to provide drastic rate reduction (e.g., 8 kb/s for G.729, 5.3 and 6.3 kb/s for G.723.1). The reduced bandwidth utilization is at the expense of additional complexity and encoding delay as well as slightly lower quality. Further reduction in the data rate can be achieved by means of Voice Activity Detection (VAD), in which case nosignal is encoded during silence periods. When silence suppression scheme is employed, the codecs then operate in two states: a silent state at zero bit-rate and an active state at the compressed bit-rate. Regardless of the state, the frame period and frame size are still fixed. After the coding operation, the packetizer encapsulates a certain number of speech samples (for G.711) or a certain number of frames (for G.729 and G.723.1) into packets of equal sizes. The protocol stack used to carry the real time voice packets is RTP over UDP/IP. As the voice packets are sent over an IP network, they are subject to variable delays and network drops. Even if a lot of voice codecs can tolerate some small packet loss without severe degradation, voice traffic has unacceptable performance if long delays are incurred. It is recognized that the end-to-end delay has a great impact on the perceived quality of interactive conversations with a threshold effect around 150 ms. For intracontinental calls, the packet delay is on the order of 30 ms and for inter-continental calls the delay can be as large as 100 ms. The impact of delay on voice communication quality varies significantly with the use. For instance, long delays are not annoying in a cell phone as in a regular wired phone because of the added value of mobility. 3.E-MODEL Perceived voice quality is typically estimated by the subjective Mean Opinion Score (MOS), an arithmetic average of opinion score that ranges from 1 to 5. Objective quality scores can be generated by comparing the impaired voice signal with its original version such as in Perceptual evaluation of speech quality (PESQ). However, PESQ do not consider the effect of delay on voice communications and neither MOS nor PESQ can be used for real-time on-line quality. The E-model is a computational model for use in transmission planning, hence a transmission rating model that can be used to help ensure that users will be satisfied with end-to-end transmission performance. The model integrates in the rating value R, called transmission rating factor (R-value), the impairment factors that affect

2 communication equipment, including delay and low bit-rate codecs. These impairments are computed based on a series of input parameters for which default values and permitted ranges are specified. These should be used if the corresponding impairment situation occurs. The general formula is: R=R 0 I s I d I e eff +A where: R 0 = basic signal-to-noise ratio I s = factor for impairments that are simultaneous with voice transmission I d = delay impairment factor I e-eff = packet-loss-dependent effective impairment factor A = advantage factor (system specific) Since the computation of the rating factor R involves a large number of parameters, complementary recommendations and appendices have been proposed by ITU-T, such as [G.108] and [G.113] that give the values for these parameters for predetermined conditions for which the model has been calibrated. 4.WHY IEEE802.11E The widespread of multimedia data and applications transmission over wireless LAN has made it necessity to a QoS support for the IEEE standard. Therefore, IEEE task group has created a special version, which is e, which adds a set of QoS enhancement to the original MAC. The aim of QoS in wireless networking is to provide priority including channel bandwidth, controlled and bounded jitter and delay (required by some real-time applications such as video streaming and VoIP), minimize the probability of losing and dropping packets. However, providing priority for specific stations to transmit does not mean that the other stations will be ignored. 5.OVERVIEW OF HCCA AND EDCA In the e standard, a super frame consists of two phases: a contention period (CP) and a contention free period (CFP). EDCA is used only in the CP while HCCA can be used in both periods. Therefore a CP consists of controlled access periods (CAP), which refer to HCCA activity, alternating with EDCA activity. HCCA is based on polling and is controlled by a hybrid coordinator (located in the QAP). In order to be included in the polling list of the HC, a QSTA must send a QoS reservation request using a special QoS management frame, and each individual traffic stream (TS) needs one reservation request. The QoS management frame contains traffic specification (TSPEC) parameters. The mandatory TSPEC parameters include the mean data rate of the corresponding application, the MAC Service Data Unit (MSDU) size, the maximum service interval (the maximum time allowed between adjacent OPs allocated to the same station) and the minimum PHY rate (the minimum physical bit rate assumed by the scheduler for calculating transmission time). Basically, each TS first sends a QoS request frame to the QAP. Using these QoS requests, the QAP determines first the minimum value of all the maximum service intervals required by the different TSs that apply for HCCA scheduling. Then it chooses the highest sub multiple value of the e beacon interval duration as the selected service interval (SI), which is less than the minimum of all the maximum service intervals. Thus, an e beacon interval is cut into several SIs and QSTAs are polled accordingly during each selected SI. EDCA defines a DCF-like random access to the wireless channel through access categories (ACs). At any node, the incoming traffic is mapped to one of the four ACs. Each AC executes an independent backoff process to determine the time of transmission of its frames. The backoff process is regulated by four configurable parameters: minimum contention window(cwmin), maximum contention window (CWmax), arbitration inter frame space (AIFS), and transmission opportunity (OP) limit. 6.QUALITY OF SERVICE PROVISIONING IN IEEE E: THE EDCA FUNCTION Like the Distributed Coordination Function (DCF), EDCA is based on the CSMA/CA scheme and employs the concept of Inter Frame Space (IFS) as well as the backoff mechanism; furthermore it introduces the following innovations. When an e station seizes the channel, it is entitled to transmit one or more frames for a time interval named Transmission Opportunity (OP); a OP is characterized by a maximum duration, called OP Limit. Various Access Categories (ACs) are defined, each of which corresponds to a different priority level and to a different set of parameters to be used for contending the channel. In particular, an e station operating under the EDCA function includes up to four MAC queues; each queue corresponds to a different AC and represents a separate instance of the CSMA/CA protocol. A queue employs the following parameters to access the channel: (i) the Arbitration Inter Frame Spacing (AIFS AC), similar to the DIFS used in DCF, (ii) the Minimum and the Maximum Contention Window (CWmin[AC], CWmax[AC]), (iii) and the OP Limit[AC]. The higher the AC priority is, the smaller the AIFS[AC], CWmin[AC] and CWmax[AC]are. The larger the OP Limit[AC], the greater the share of capacity of the AC. However, the values of CWmin[AC] and CWmax[AC] have to be carefully chosen so as to avoid high collision probability among traffic flows belonging to the same AC, and the value of AIFS must be at least as long as the DIFS interval (the only exception is for the AP that can use an AIFS 30 long). Within every e station, a scheduler solves virtual collisions among the AC queues, i.e., among the various CSMA/CA instances, by always enabling the queue associated with the highest priority to transmit. 7.ANALYSIS OF IEEE E QOS ENHANCEMENTS The original IEEE standard specifies two channel access mechanisms: a mandatory contention-based distributed coordination function (DCF) and an optional polling based point coordination function (PCF). DCF provides a best effort service and is not capable of providing differentiation and prioritization based upon traffic type. While DCF may provide satisfactory performance in delivering best-effort traffic, it

3 Techniques to Improve Performance of VoIP over e WLAN lacks the support for QoS requirements posed by real time traffic, and especially VoIP which has stringent delay requirements. These requirements make the DCF scheme an infeasible option to support QoS for VoIP traffic. PCF mode, B OP1 OP2 ED CA OP OP1 OP2 Fig. : IEEE e Superframe with a centralized controller, represent another promising alternative to providing QoS in WLAN. Nevertheless, studies on carrying VoIP over WLAN in PCF mode in found that when the number of stations in a basic service set (BSS) is large, the polling overhead is high and results in excessive end-to-end delay and that VoIP still get poor performance under heavy load conditions. Thus, neither DCF nor PCF presents sufficient functionality to provide the QoS demanded by multimedia applications. In WLANs, the physical layer's error rate is larger than that of wired LAN and the challenges of the wireless channel make physical layer data rate improvements difficult to achieve. These reasons have led to the development of service differentiation based MAC schemes that classify traffic types based on their relative priorities. The IEEE e standard addresses the shortcomings of the standard, defines a superset of features backward compatible with DCF/PCF and introduces the Hybrid Coordination Function (HCF). HCF has two modes of operation: Enhanced Distributed Coordinated Access (EDCA) and HCF Controlled Channel Access (HCCA). EDCA inherits all the contention scheme and parameters of the original DCF and provides service differentiation through prioritized access to the wireless medium. Prioritization is realized through the introduction of four Access Categories (ACs) each with its own transmit queue and set of AC parameters. The differentiation in priority between ACs is realized by setting different values for the AC parameters which include the arbitration inter frame spacing (AIFS) and minimum contention window size (CWmin). With proper tuning of these parameters, the performance of delay sensitive multimedia traffic can be improved. In an infrastructure network, the AP will again access to the medium with a higher priority than other QoS Stations (QSTAs). Under HCF the basic unit of allocation of the right to transmit onto the wireless medium is the transmission opportunity (OP). Each OP is defined by a starting time and a defined maximum length. The OP may be obtained by a QSTA winning an instance of EDCA contention during the CP, or by a non-ap QSTA receiving a QoS + CF-Poll during the CP or CFP. With the HCCA, a hybrid coordinator (HC) allocates transmission opportunities (OPs) to wireless STAs by polling, so as to allow them contention-free transfers of data, based on QoS policies. QSTAs may obtain OPs using one or both of the channel access mechanisms. An HC generates an alternation of CFP and CP, the sum of the two periods forms ED CA OP B the superframe (SF). In addition, contrary to DCF, QSTAs can be polled during thecp in periods called Controlled Access Periods (CAPs). The duration of OPs allocated to each QSTA is determined by the HC schedular according to the requested QoS parameters. It is also possible to change the SF length since beacons carry a parameter indicating the SF length. Due to the service differentiation, the real-time traffic gets a higher priority in winning channel contention under e and evidently provides a better performance as compared to the basic MAC scheme. 8.PROBLEMS ON EDCA As described above, EDCA provides prioritized QoS for different ACs. However EDCA does not take into consideration the QoS of traffics which belong to the same AC. The packet collision probability increases as the number of calls increases due to simultaneous transmission. Additionally the delay tends to increase if more than one STA are in backoff states. These problems result in voice quality degradation. Therefore, it is necessary to restrict the number of VoIP calls over a WLAN system. Instead of EDCA, IEEE802.11e also provides the polling based MAC protocol called Hybrid coordination function Controlled Channel Access (HCCA). However HCCA proposes no concrete scheduling algorithms. In addition, collision of polling packets from more than one access point (AP) may always occur in multiple-cell environment because VoIP application generates packets to transmit at a constant interval. 4 PROPOSED TECHNIQUES 9.OVERVIEW OF PROPOSED SCHEME We propose a MAC protocol to provide effective QoS guarantees for VoIP over WLAN. Our proposed protocol follows three policies: (1) No modification of access points, (2) No hardware (H/W) modification of VoIP terminals, and (3) Maintain backward compatibility. The first policy enables users to implement the proposed method on existing APs. The second one minimizes the impact on implementing the proposed method in VoIP terminals. The third one prevents the proposed method from blocking communications among terminals that are equipped with only conventional techniques. After each STA connects to AP, each STA decides its own transmission timing using the distributed transmission scheduling. We call this stage the observation phase. After that, the scheduled transmission phase follows. In the scheduled transmission phase, each STA transmits packets using the dynamic priority setting. It is also possible to re-schedule transmission timing if necessary. Our proposed techniques are premised on APs and STAs which host Unscheduled Automatic Power Save Delivery (U-APSD), which is defined in the IEEE802.11e. The U-APSD is intermittent reception mechanism that AP transmits a packet upon receiving a packet from an STA. 10.DISTRIBUTED TRANSMISSION SCHEDULING

4 Because VoIP application generates packets to transmit at a constant interval, each STA has only to decide a schedule in a certain VoIP codec period. Each STA in single-cell environment proceeds as follows. First of all, in order to detect other STAs, each STA observes down-link packets from AP and reads the destination address field in the MAC header. This enables each STA to recognize others. Secondly, each STA makes a list of MAC addresses of STAs which belong to the same AP. The final step in this phase is to sort the MAC addresses in ascending order and make a scheduling table. This enable all the STAs in the same cell to share a scheduling table. During this phase, STAs transmit packets using EDCA. 11.DYNAMIC PRIORITY SETTING After the scheduling table is made, each STA creates a virtual slot periodically. Each STA sets the priority of its virtual slot according to the scheduling table. Specifically, each STA dynamically changes its AIFSN, CWmin and CWmax. The virtual slot of a VoIP STA is shifted to the next VoIP STA in the length of the time required for transmission of up-link and down-link packets. Moreover, the duration of a virtual slot is longer than the length of the time required for transmission of up-link and down-link packets. Therefore, the virtual slot of a VoIP STA partially overlaps with that of the next STA. This leads to flexible scheduling which prevents the scheduled sequences from collapsing due to interrupts from the previous VoIP STA or STA equipped only with conventional techniques. If the scheduled sequences collapse due to join of new STA, each STA adopts any of the following methods. One way is for the new STA to transmit a re-scheduling request packet in a broadcast manner. The other is for any STA connected already to detect frequent failures of reception of an acknowledgement (ACK) from AP and then transmit a rescheduling request packet in a broadcast manner. Figure 2. Multiple-cell environment 12.APPLICATION TO MULTIPLE-CELL ENVIRONMENT The number of the available frequency channels is very limited in IEEE WLAN. This is why it is inevitable that more than one cell with the same frequency channel will overlap. In this section we extend the proposed techniques to implement the distributed transmission scheduling for multiple-cell environment. In Figure 2. AP1 and AP2 use the same frequency. VoIP A, B and C are connected to AP1 and VoIP D and E are connected to AP2 and VoIP C lies in the overlapping area of AP1 and AP2. There is a possibility that the virtual slot of VoIP C will overlap with that of VoIP D and E. This leads to packet collision between VoIP C and VoIP D/E due to the transmission at the same time. Hence the virtual slots of the VoIP STAs in an overlapping area need to be set in different timing from those of VoIP STAs in not only the same cell but also neighbor cells. At the beginning, each VoIP STA behaves in the manner and makes the scheduling table of VoIP STAs which belong to the same AP. After that, each VoIP STA determines if packets from other APs are being received. This enables VoIP C to find itself in an overlapping area. VoIP C searches the virtual slots which are not occupied by the VoIP D/E in neighbors cell. Then VoIP C transmits a slot request packet including the virtual slot number which is not occupied by the VoIP D/E in a broadcast manner. This leads for VoIP A and B to receive the necessary information and make the scheduling table under the multiple-cell environment. CONCLUSIONS This paper presented a MAC protocol that provides effective QoS for VoIP over WLAN. The characteristics of our proposed protocol are (1) No modification of access points, (2) No H/W modification of VoIP STAs and (3) Backward compatibility in order to inimize the costs of development and introduction.. 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