Fathelalem F.Ali Meio University, Nago, Okinawa , Japan Tel: (81) Fax : (81)

Transcription

1 Performance Evaluation of VoIP over Wireless LAN Limbran Sampebatu Meio University, Nago, Okinawa , Japan Tel: (8)98-5- Fax : (8) lsampebatu@yahoo.com Fathelalem F.Ali Meio University, Nago, Okinawa , Japan Tel: (8)98-5- Fax : (8) ali@mis.meio-u.ac.jp Track 2: Parallel and Distributed Computing and Networks: Computer Networks and Communications

2 Performance Evaluation of VoIP over Wireless LAN ABSTRACT In this research we assess the suitability of 82.b networks to carry real time voice traffic communication using the Internet Protocol, where in the network the voice sessions have to share the link with data, video, and voice. Measurements are performed in the 82. wireless networks as well as wired. Measurements are focused on the three major VoIP parameters: loss, delay, and jitter. The environment factors such as the distance between nodes, network traffic, and other obstacles are also considered. Keywords: voip, loss, delay, jitter,wlan.. Introduction As we know, when deploying VoIP in this environment, one of the challenges is meeting the performance requirements to provide a best condition. When roaming around in a wireless network, at some point the mobile device will have to switch access point being associated with because the media in wireless network is not constant. These media maybe very different for each type of best condition and each of it will face an added effect to the delay, jitter, losses as a network problem that effecting the voice quality. Voice quality is an end-to-end matter with various network equipment, protocols, and policies. The performance of voice service is sensitive to delay, and the voice quality can tolerate some dropped packets within a certain limit of packet loss rate. It is supposed that delays are made small enough to make two ways conversation possible. Beside that, it is important to know how long it takes for a packet to travel through a network. The quality of a voice call is not only depends on network conditions, but also on the perceptual characteristic of the end users and how these network impairments affect the user`s perception. Other factors are the user`s expectation, since the mood and human memory can increase or decrease the perceived quality. 2. Objectives We conduct measurements of critical parameters that affect the quality of voice over wireless networks, namely loss, delay, and jitter. Delay has the important characteristics that once introduced it cannot be hidden or minimized, hence it is important to know where exactly it is introduced into the system. Loss on the other hand can be compensated for typically by adding redundancy to the transmitted signal allowing some of the received signal to be reconstructed at the receiver. [2] These researches explore the causes of the degradation of those parameters, and find an optimal condition of the environment for software, hardware to be deployed in order to improve voice performance of VoIP over WLANs. 3. Methodology and Tools To analyze the suitability of IEEE 82.b networks to carry voice traffic we performed a series of experiments covering a range of factors affecting the voice quality, including both networks and link quality aspects. To compute the end-to-end delay, loss, and jitter at the end point we used tools as mention below:

3 3.. SJPhone We used SJPhone, a soft phone with low delay communication tool. It implements two basics modules: a sender and a receiver. [3] 4. Ethernet to Ethernet 3.2 QoS Measurement Tools QoS measurement is a central component for testing the quality voice data in a wireless VoIP Communication network. Following is the short description of the software tools used for measuring the QoS parameters: 3.2. ClearSight Analyzer ClearSight is network analyzer software that completely overlooked on everyday traffic and issues in the network. Designed for packet decoding and network diagnosis, monitors the network traffic transmitted over a local host and a local network with the ability of real time packet capture and the network troubles. ClearSight Analyzer was installed in the receiver or sender nodes through a hub to be connected to the network to capture all the IP packets and MAC frames. [4,5] Ethereal Ethereal allows to monitor both MAC layer frames and IP layers traffic and that it is available under an open sources license. The fact that Ethereal has a large community supporting to its development. Ethereal was used in the receiver and sender nodes to capture all the IP packets and MAC frames. [6] Fig 4. Ethernet to Ethernet In this scenario we connected the transmitter and receiver sides through a conversation, and we captured data during the conversation. The following are the observations on loss packets and the average jitter per call Tx(Wired) Rx(Wired) Fig 4.. (Ethernet to Ethernet) Jitter Tx(Wired) Jitter Rx(Wired). 4. Experiment Scenarios For this experiment we have three experiment scenarios: Fig. Jitter (Ethernet to Ethernet) Assuming the ideal conditions on the Mbit Ethernet a setup delay at this level is not noticeable to the user.

4 4.2 Ethernet to Wireless have an increasing percentage of losses that is not followed by a higher jitter or delay. For that losses have no significant impact on the delay and jitter since loss packets are simply ignored when calculating the delay and jitter value. 4.3 Wireless to Wireless Fig. 2. Ethernet to Wireless From the above experiment environment, below are the data captured during the conversation: Tx(Wireless) Rx(Wired) Fig4.2. (Wireless to Ethernet) Jitter Tx(Wireless) Jitter Rx(Wired) Fig4.2.2 Jitter (Wireless to Ethernet) Fig3. Wireless to Wireless In these experiments we were interested in the effect of the obstacles may have in the voice quality. The experiments were done within Meio University, one node was fixed in one place and the other node was moving around represent the different receiver locations Jitter Tx(Wireless) Jitter Rx(Wireless) Fig4.3. (Wireless to Wireless) The percentage of losses exceeds the limit (5%) several times we defined for codecs that deliver an acceptable quality level. The jitter and delay values are small comparing to the suggested maximum values accepted as good. One interesting result is that jitter and delay don`t always increase with the percentage of losses. In this case we Fig3. Jitter (Wireless to Wireless) Tx(Wireless) Rx(Wireless)

5 5. Results and Discussion In this research experiment, we find out that degradation of those parameters mostly caused by the network condition during the conversation, the traffic and distance of each node to access point. In the normal condition, the closer it is to the access point the stronger the signal will be received. The losses of packet lead to the loss of some word during conversation., while jitter lead to non sequential packet arrival, and if it isn`t possible anymore to restore the original packet sequence, and since the packets will never be combined in a wrong sequence on the receiving side, such packets (although arrived) also lead to losses in speech. Delay is affecting the speech quality, but it is always present, no matter how bad the connection is, it only gets longer or shorter depending on the radio connection quality. But it never exceeds the range of about 5 ms- ms. The delay that exceed is an accumulative time of: ) Network delay 2) Queuing delay 3) Speech encoding/ decoding delay 4) Compression / decompression delay 5) Transmission delay Depending on the codec utilized to transform the analog signal into a digital stream of packets, the percentage of lost packets must be kept under a certain minimum; ) If the loss in the range of %- %, the call isn`t seriously disturbed, from about % the loss is noticeable. 2) If the loss between %-2%, the conversation is exist, but the repetition will be asked oftenly to understand the other part. 3) If the loss beyond the 2% mark, a conversation is not possible anymore. of Internet Protocol has to be in a good condition, the number of access points are enough to cover the whole area in the network to assure that all mobile nodes are associated to it. Beside that, there are some demands for some quality of service from the network. The voice stream must not suffer a delay higher than 5 ms (including processing delays added at the end systems plus the network latency) because this would lessen the interactivity of the conversation. 6. VoIP in the future As the Internet growing fast recently, VoIP just like PCs, web access, will grow out of the enterprise market and get into the residential market. For this technology is cheaper, simpler, and more convenient to use it is expected to replace the legacy network infrastructure. However the appeal of the efficiencies of VoIP and Wireless Network (82.) cannot be ignored and the end result may be very well be a mix of technology. In our future research, we will explore the possible applications of wireless IP phone in many areas of our daily life such as hotels, hospitals, schools. References: [] Frank Ohrtman,Voice over 82., Artech House,Inc, USA, 24 [2] William C.Hardy, VoIP Service Quality, McGraw-Hill, USA, 23 [3] [4] oduct.html, [5] ing/cs_ope.html [6] ( In order to have the best conversation of VoIP through 82. wireless network, the environment as one of the backbone instead