VOIP on Wireless LAN: A Comprehensive Review

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1 VOIP on Wireless LAN: A Comprehensive Review Sana Munir [1], Imran Ahmad [2] [1] Center for Advanced Studies in Engineering,Islamabad, PAKISTAN [2] NWFP University Of Engineering and Technology, Peshawar, PAKISTAN Abstract VoIP requires timely servicing of the voice traffic, which is a challenging task in WLANs, even when using QoS enforcement. The use of Wireless LANs (WLANs) is more and more at present. Voice over IP (VoIP) uses data networks to transmit voice signals. Since both technologies are sufficiently established at the moment, VoIP over WLAN communication is being developed. However the inherent characteristics of each of these two technologies cause specific issues to appear. These issues must be addressed in order to ensure a successful deployment of VoIP over WLANs. This is particularly important when using WLAN technology in the context of emergency situations. A review of the current state of the art in voice communication over wireless networks is done in this paper. The properties of WLANs and VoIP are presented, and then the issues related to the deployment of VoIP over WLAN are analyzed. Keywords: Wireless LAN, VOIP, IEEE e, RSVP, QoS 1. INTRODUCTION: The emergence and continual growth of wireless local area networks (LANs) are being driven by the need to lower the costs associated with network infrastructures and to support mobile networking applications. They also represent a solution for the creation of ad-hoc networks in emergency conditions within areas where dense wireless networks exist. Most wireless LANs operate over unlicensed frequencies (11 Mbps or 54 Mbps) using carrier sense protocols to share a radio wave or infrared light medium. These rates are considerably lower that the current extensively-used 100 Mb/s and 1 Gb/s fixed LANs. The significant amount of management and control traffic, plus contention for the radio frequency spectrum, constitutes the overhead. Additional features, such as encryption, increase even more the overhead and diminish the good put of the system. That is why b equipment, which can in theory run up to 11 Mb/s, have shown that in practice the sustained rate only climbs up to about 6-7 Mb/s [5]. Wireless LANs perform functions similar to their wired Ethernet. In wireless networks the single medium is used by the applications of all the users, hence the network quality is more degraded as compared to the wired networks. In WLANs where more access points are simultaneously active, another issue is roaming. When a node moves or reception conditions change, it will usually select the access point in its range that has the highest signal strength. Roaming is the event of switching from one access point to another which causes a significant amount of delay. This delay is unacceptable in real time applications. Every transmitting and receiving node in the network, such as the AP in a WLAN, typically buffers incoming and outgoing packets. This causes latency and jitter. The overhead of roaming, security mechanisms, retransmissions, as well as data and voice convergence add latency and jitter in WLAN. Voice over IP (VoIP), also known as Internet telephony, is a form of voice communication that uses data networks to transmit audio signals. When using VoIP the voice is appropriately encoded at one end of the communication channel, and sent as packets through the data network. After the data arrives at the receiving end, it is decoded and transformed back into a voice signal. It has two fundamental benefits compared with voice over traditional telephone networks. First, it can improve bandwidth efficiency by using advance voice compression techniques and bandwidth sharing in packet switched networks. Second, it facilitates the creation of new services that combine voice communication with other media and data applications like video, white boarding and file sharing. ISSN: Page 225 ISBN:

2 This paper is a survey of the current state of the art in voice communication over wireless networks. Due to the nature of wireless networks and its aforementioned characteristics, specific issues appear that must be addressed in order to ensure a successful deployment of VoIP over WLANs. The paper is structured as follows. First we introduce the most widely spread WLAN standards, with emphasis on networks and their recently added QoS features,in section 2 In section 3 we present VoIP telephony and the particular issues related to it. The section will discuss VoIP over WLAN in the light of previously provided information and certain factors which inhibits the performance of WLAN for VoIP applications. At the end conclusion summarizes the main findings of this review. 2. WIRELESS LANs There are several wireless LAN specifications and standards that you can choose from when developing wireless LAN products. The different IEEE standards are given below. The a standard defines operation at up to 54 Mbps in a 5 GHz band using orthogonal frequency division multiplexing. The b defines operation at up to 11 Mbps in an unlicensed band 2.4GHz using DSSS-CCK modulation. The g defines operation up to 54 Mb/s in the 2.4 GHz band, using OFDM or DSS with CCK modulation. At the moment b is probably the most widely used WLAN standard, but there are devices that are compatible with all three standards in the same time. As always in the ITC the tendency is to migrate to faster technologies as soon as they become affordable. Hyper LAN is the European standard. It is available with two standards and it operates up to 20 Mb/s and 54 Mb/s, HiperLAN/2 provides better QoS than HyperLAN/1, and bandwidth guarantees. The IEEE family (WiMAX) is a specification for fixed broadband wireless metropolitan access networks (MANs) with a bandwidth of up to 75 Mb/s. It operates in the GHz range (with support for 2-11 GHz for the a variant). WiMAX uses OFDM modulation and DES37 & AES8 security. Features like Quality of Service (QoS) establishment on a per-connection basis, strong security, and support for multicast and mobility are being added to WiMAX as well. This technology is currently used to interconnect local WLAN clouds over larger distances, hence extending significantly the potential coverage area of a wireless network. Bluetooth is another wireless technology, which can deliver up to 2 Mb/s in the 2.4 GHz band. It uses FHSS9 modulation and PPTP10, SSL11 or VPN12 security. Bluetooth offers point-to-point links and has no native support for IP; therefore it doesn't support TCP/IP and wireless LAN applications well. Bluetooth is best suited to connect PDAs, cell phones and PCs in short intervals. IEEE e The current Quality of Service (QoS) standard for wireless networks from the widely-used e family is IEEE e, which has been approved and published in November 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. In addition the standard provides improvements in the capabilities and efficiency of the protocol. These enhancements, in combination with the improvements in PHY capabilities of a and b, are expected to increase overall system performance, and expand the application space for Example applications include transport of voice, audio and video over wireless networks, video conferencing, and media stream distribution. Original MAC protocol The legacy PCF was primarily designed to provide limited QoS support to multimedia delivery in WLAN.[2] In this section, some of the limitations of the PCF are reviewed which led to the development of the enhanced e protocol. In infrastructure mode both PCF and DCF coexist and in this case time is divided into super frame. Each super frame consists of a CFP and CP. PCF is used in contention free period. During the CFP, the AP sends poll frames to stations when they are clear to access the medium. The boundaries between CFPs and CPs are marked by beacons carrying the Delivery Traffic Indication Message (DTIM), which is used to wake up stations in power-save mode to receive any buffered data frames. Mobile hosts can use the information present in the beacon frames in order to associate with the AP, which is performed during the CP. This association is necessary if the terminal needs to transmit any data on the network and have its transmissions scheduled by the PCF, which is usually required for QoS sensitive data. However the current PCF faces certain limitations which hinder its ability to support QoS.First the unpredictable beacon delays and unknown transmission durations of the polled stations in PCF mode. Second, stations can start their ISSN: Page 226 ISBN:

3 transmission even if the MAC service data unit (MSDU) delivery cannot finish before the upcoming target beacon transition time (TBTT). Third, the unknown transmission time of polled stations. A station that has been polled by the PC is allowed to send a possibly fragmented single frame and of arbitrary length. Finally, different modulation and coding schemes are specified in a, thus the duration of the MSDU delivery that happens after polling is not under the control of the PC. As such, further QoS support to other stations that are polled during the rest of the CFP is not possible. QoS enhancement in e In this subsection, the e enhancements EDCA and HCF are discussed. The legacy DCF cannot support the QoS requirements of multimedia applications. The proposed solution EDCA includes a prioritization enhancement based on different Access Categories (ACs). One or more user priorities (UPs) can be assigned to each AC. Each AC has a unique queue, IFS (Arbitration IFS, AIFS) and contention window parameters. Each AC contends for medium access with only one CSMA instance using the parameters that belong to its lowest UP. This corresponds to the priority of the AC as a whole [2]. Besides EDCA, e QoS facility includes an additional coordination function called HCF. Like PCF, HCF includes a polling mechanism which is controlled by AP which is used during controlled Access Periods.(CAPs).HCF is more flexible than PCF in sense the CAPs can occur anytime during the super frame, with the ratio between contention free and contention transmission being controlled by a token-bucket of time units. Whereas in legacy PCF, contention free period (CFP) has a fixed position in the super frame. The QoS sensitive traffic has to wait for the entire DCF contention period before being polled. During the CP, the HCF employs the DCA mechanism that provides differentiated, distributed access to the WM for 8 UPs for QoS stations. The QoS facility accepts traffic from higher layers that belong to one of eight possible UPs. The frames will then be mapped from to one of four possible ACs. Collisions between ACs within a single station are dealt with internally and this case is referred to as virtual collision detection.. To start a CP period, the same rules applied in EDCA are used. There are several factors that impede WLAN performance which are: Excessive latency and jitter, leading to degraded voice quality Poor coverage Roaming latency between Access Points (APs), leading to interrupted voice service Security issues Retransmissions and dropped packets Low capacity, reduced number of calls Quality of service, required for voice and data convergence Power consumption requirements 3. VoIP: The audio signal from an input device such as microphone is transformed into digital form by an analogto-digital converter. The voice data is then packetized and encoded. Encoding (as well as decoding) is done by codecs that transform sampled voice data into a specific network-level representation and back. After binary information is encoded and packetized at the sender end, packets encapsulating voice data can be transmitted on the network. Voice packets interact are routed through shared medium (used by data packets as well) to their destination. At the receiver end they are decapsulated and decoded. The flow of digital data is then converted to analogue form again and played at an output device, usually a speaker. Issues: Since the communication channel is not reserved but shared with other applications, voice packets can arrive at the receiver with a different inter-packet gap than they had at the sender, out of order, and some of them can even be lost. [2] Assessing the relationship between precisely these factors, as quantified by means of network QoS parameters, and the User-Perceived Quality (UPQ) of VoIP communication is a prerequisite for any performance and dependability analysis of VoIP over WLAN. Let's analyze the influence of each of the main QoS parameters now. Given the low requirements of VoIP in terms of bandwidth (64 kb/s maximum), bandwidth in usually not a problem, at least for individual voice calls. Simultaneous voice calls however can have a cumulated throughput requirement that approaches the limits of the network equipment used. Delay & jitter are probably the most important for VoIP as a real-time streaming application. Packets containing voice data must be delivered in a timely manner in order to ensure user satisfaction. One-way delay influences interactivity: the larger the delay the lower the perceived interactivity for the persons who communicate. Jitter on the other hand (i.e. one-way delay variation) influences quality if it exceeds a maximum value. This maximum value is system dependent, and is related to the size of the de jittering buffer used. A large ISSN: Page 227 ISBN:

4 buffer means that jitter has a smaller effect on perceived quality, but it decreases interactivity through the effect of delay. If the induced jitter value exceeds the size of the dejittering buffer, then VoIP packets don't arrive in time for playback, and playback signal quality drops. Hence this distortion is the main effect that jitters has on user satisfaction. Packets that don't arrive in time for playback can be considered lost; therefore this effect is sometimes termed jitter-loss. VoIP data streams are usually of UDP type, and in this case packet loss has only momentary effects. When packets are missing for playback, the system either introduces gaps in playback, or tries to recover from this error by replacing the gap with something more appropriate (previous voice samples, a reconstructed signal, etc.). No matter what the system's robustness is, packet loss will surely cause a certain quality degradation to occur. This degradation is larger when loss happens in bursts (a number of consecutive packets being lost) since such an event has a higher influence on VoIP perceived quality than spaced losses. Unfortunately bursts are precisely how losses occur in real networks, since congestion will hardly ever affect only one packet at a time. VoIP quality measurement In order to evaluate system performance when using various applications it is necessary to use specific metrics for each application; this makes it possible to measure the User- Perceived Quality (UPQ) for the corresponding application in an objective manner [2]. ITU-T has defined several standards that allow an evaluation of the quality of voice communication. The first of them was a subjective metric (the MOS), but successive attempts have been made to define objective metrics as well. The two main methods to measure the VoIP UPQ that are currently used are the E-model, and the PESQ score. Mean Opinion Score (MOS) The following values are assigned depending on the quality of the connection: Excellent=5; Good=4; Fair=3; Poor=2; Bad=1 The distinction between mean conversation-opinion score (MOSC) and mean listening-opinion score (MOSL) is made. In the second case only the intrinsic audio quality is taken into account, whereas the first case includes the experimenter's opinion about the level of interactivity. E-model The model integrates in the rating value R, called transmission rating factor (R-value), the impairment factors that affect 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= R0 Is Id Ie eff-a where: RO= basic signal-to-noise ratio Is= factor for impairments that are simultaneous with voice transmission Id= delay impairment factor Ie= 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 pre-determined conditions for which the model has been calibrated. PESQ PESQ score computation requires both the original and the degraded voice signal. The key process in PESQ is the transformation of both the original and degraded signal into representations analogous to the psychophysical representation of audio signals in the human auditory system. The PESQ score is mapped by design to a MOS-like scale, a number in the range of -0.5 to 4.5, although for most cases the output range will be between 1.0 and 4.5, the normal range of MOS values found in subjective listening quality experiments. The relationship between PESQ scores and audio quality is the following: PESQ scores between 3 and 4.5 mean acceptable perceived quality, with 3.8 being the PSTN22 threshold this will be termed as good quality; Values between 2 and 3 indicate that effort is required for understanding the meaning of the voice signal this will be named low quality; Scores less than 2 signify that the degradation rendered the communication impossible, therefore the quality is unacceptable. Unlike the E-model, it doesn't require any knowledge regarding the network and uses only the original and degraded signal to compute the PESQ score. ISSN: Page 228 ISBN:

5 Metric comparison Since MOS is not suited for automated testing, we'll focus in this section on a comparison between the two objective metrics, the E-model and the PESQ score. For PESQ score computation voice recording capabilities are essential in order to have an accurate estimate, whereas for the E-model's R-value it is mandatory to make traffic measurements and appropriately choose the values of the model parameters. The comparison between the two can be summarized in a table. 4. VoIP OVER WIRELESS LANs IP telephony has low-bandwidth requirements (below 64 kb/s); however combining the two technologies today is difficult. Experiments show that even a small amount of data traffic on the same network can lead to seriously degraded audio quality and dropped calls, even with QoS features enabled [Net-05]. When voice and data traffic are sent on the same network, contention must be managed in terms of delay & jitter rather than forwarding rates. Performance of VoIP (and real-time applications in general) over wireless media and the difficulty in finding appropriate QoS solutions depends upon the inherent properties of WLANs. In wireless networks packet error rates can be in the range 10-20%. Moreover bit rates vary according to channel conditions; hence bandwidth reservation at connection setup time might not hold throughout the entire duration of the communication. Voice over WLAN Challenges Excessive Latency and Jitter, Degraded Voice Quality: One of the major differences between VoIP applications and data applications is the sensitivity to latency and jitter [6]. Every transmitting and receiving node in the network, such as the AP in a WLAN, typically buffers incoming and outgoing packets. This causes latency and jitter. The overhead of roaming, security mechanisms, retransmissions, as well as data and voice convergence add latency and jitter in WLAN. Roaming Latency; Interrupted Voice Service: Traditional WLAN topology uses cell planning to achieve coverage. As mobile users move from cell to cell, inter- AP handoffs are required. A handoff requires several steps including AP discovery, re-association, security measures, and higher-level protocol exchanges which introduce latency and jitter. AP discovery typically takes milliseconds, and introduces milliseconds of jitter in b and g. AP discovery latency and jitter in a or in multi-mode networks may be two times greater or more. In traditional topologies, inter-ap handoffs are controlled by the clients. Clients require a long time to select an AP, and tend to remain with an AP, even if it is far away. Not handing off quickly enough or handing off to a less than optimal AP causes rate adaptation, which introduces more latency and jitter, reduces capacity, and increases power consumption. The voice quality is also degraded as the packets buffered in previous AP may be dropped. Newer solutions, which provide AP topology awareness, only marginally improve the described latency and jitter. Security Issues: Wired Equivalence Privacy (WEP), the original security protocol for wireless networks, is now considered flawed by the industry. The IEEE i standard introduced newer security standards for WLAN. The interim WiFi Protected Access (WPA) includes temporal key encryption (TKIP), message integrity checks (MIC) and strong authentication (EAP), thus resolving several of the security problems of WEP. While WPA is necessary to provide adequate security but it introduces an additional overhead at each inter-ap handoff. Voice without security is unacceptable for an enterprise deployment. Retransmissions and Dropped Packets: Wireless medium is notoriously unreliable, and therefore the WLAN standard retransmission mechanism is introduced to avoid dropped packets. This is a good solution for data but it results in a significant amount of jitter and latency for voice applications. To abandon retransmissions in voice applications, a low drop rate is required which is impossible with traditional topologies. In traditional topologies, large distances from the AP, temporary and random interference and obstructions, as well as cochannel interference caused by neighboring cells all cause poor reception of packets. Radio Frequency (RF) technologies at neighboring organizations or sites may add another source of interference. Unlike other causes of poor packet reception, neighboring sites cannot be controlled or overcome by a more careful deployment. Without retransmissions, a large drop rate and consecutive dropped packets occur. With retransmissions, significant latency and jitter is experienced, and the capacity is greatly reduced. In either case, voice quality is severely degraded. Low Capacity, Reduced Number of Calls: Although the major requirement for voice is low latency and jitter, but adequate capacity is also required. Capacity determines ISSN: Page 229 ISBN:

6 the number of concurrent calls that can be supported. The measure of capacity for voice applications is the number of packets per second (PPS) that can be transmitted. Several factors limit the PPS of a WLAN, the most important of which are contention windows, acknowledgment (ACK) packets, retransmissions, and rate adaptation. The proposed IEEE e standard reintroduces contention-free access (scheduled access), termed the Hybrid Coordination Function (HCF) similar to the original Point Coordination Function Contention-free access provides a higher PPS, resulting in double the number of available voice calls.5 Unfortunately, PCF was never supported by mobile clients, and client support for HCF is not yet available. Multiple co-channel APs are bound to cause interference, making it difficult to use scheduled access in cell-based topologies. ACK packets are one the most significant causes of PPS reduction and can account for 30 40% of the airtime used in a single VoIP packet transfer. High drop rate results in retransmissions which increases the need of ACK packets. Furthermore, retransmissions trigger rate adaptation at the client. As a result, more airtime is required to transmit the same packet, resulting in a reduced PPS.6 This, in turn, necessitates ACK transmissions, reducing the network PPS even further. Furthermore, for a user to be able to move from cell to cell, some slack is necessary in each cell to allow a user to roam in. This means that even the already low capacity cannot be fully utilized. The restrictions described above are so severe that current recommendations suggest a maximum of 5 7 voice calls per b network. Power Consumption Requirements: Mobile users that are close to an AP are able to transmit at the highest data rate. Transmitting at the highest data rate reduces the time the transmitter is on, thus decreasing the client s power consumption.7 In addition, if the mobile unit is close enough to the AP, it can successfully transmit a packet with greatly reduced transmission power. The choice of transmission standard, OFDM (Orthogonal Frequency Division Multiplexing) or CCK (Complementary Code Keying) also affects power consumption. OFDM provides higher data rates, reducing transmission time but it requires extremely short distances from the AP and consumes more power during packet reception. Compared with OFDM, CCK is more power efficient, and provides greater distance but it supports lower data rates than OFDM. Since the distances required for voice applications cannot be achieved by OFDM in traditional topologies, and the power consumption of OFDM chips is currently too high as well, therefore, CCK is currently used. This results in a mixed g and b network, which significantly reduces network capacity. 5. CONCLUSIONS WLAN provides an excellent opportunity to enable VoIP, since it combines the cost effectiveness of VoIP solutions with cordless mobility. However, such deployments have special needs in order to be effective. This requires a very strong uplink to reduce latency and jitter, complete coverage and seamless mobility, strong security, without interrupting service with constant handoffs. In addition, increased capacity is needed to provide a sufficient number of simultaneous voice calls. Users must always be close to an AP to cope with power constraints of mobile devices, allowing them to use lower transmission powers, and to transmit at the highest data rate. REFERENCES: [1] Network World, Review: Voice over Wireless LAN, white paper, January [2] Weiwang, Soung C. Liew, Solutions to Performance Problems in VoIP over Wireless LAN [3] Rezwan Beuran, VoIP over Wireless LAN Survey [4] The International Enginee The International Engineering Consortium, Including VoIP over WLAN in a Seamless Next- Generation Wireless Environment, white paper, [5] Network World, Review: Network World Review: Voice over Wireless LAN, white paper, January [6] Voice over Wireless LAN, Belden- Sending all the right signalsnetwork World, Review: ISSN: Page 230 ISBN:

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