VoIP quality aspects in b networks. Victor Yuri Diogo Nunes

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1 VoIP quality aspects in b networks Victor Yuri Diogo Nunes September 2004

2 Abstract IEEE b is increasingly being used in public and oce networks. Therefore it must cope with both voice and data trac. In this thesis we assess the suitability of b networks to carry real-time voice trac using the IP protocol. Measurements were performed over b networks in both ad-hoc and infra-structure modes and focused on the three major VoIP metrics: loss, delay and jitter. The environmental factors such as the distance between the nodes and information about intervening obstacles were recorded wherever possible. The eect of competing trac was also analyzed. We have found that b networks can support the requirements of realtime voice communication when there is clear line of sight to the peer node or when communicating nodes are close. Distance and obstacles proved to cause loss and burstiness, whilst competing trac increased the delay leading to worse quality. All experimental data was recorded in a repository for further study.

3 Contents 1 Introduction Motivation and goals Thesis structure Background Network layers and protocols UDP protocol IP protocol IEEE b standard data link and physical layers Real-time protocols RTP RTCP Voice QoS End-to-end delay Packetisation delay: Queuing delay: Serialization delay: Propagation delay: Jitter Loss Subjective metrics Objective metrics Sister project Approach and method Approach Method Ad-hoc mode Infra-structure mode Tools Sphone Extending Sphone: the real-time tool Log-le generation Real-time metrics

4 Sphone timestamp validation Ethereal Matlab Results Ad-hoc b mode Open space measurements using line of sight Loss Delay and jitter Oce environment using line of sight Oce environment with obstacles Losses Delay and jitter Ad-hoc mode with competing TCP trac Losses Delay and jitter Infra-structure mode with competing TCP trac Losses Delay and jitter Microwave interference Output data and voice signal properties Call properties and gathered data Repository structure Problems Repeated packets in infra-structure mode Centrino cards Collisions detection with Ethereal Related work 44 6 Discussion 46 7 Conclusions 49 8 Future work 50 9 Lessons learned 52 A Glossary 57 B Measurement setup 58 B.1 Equipment B.2 Experiment setup B.2.1 Installing Sphone B.2.2 Ad-hoc experiment B Laptop setup B Running experiments

5 B.2.3 Infra-structure experiments B Laptop setup B Running experiments C Matlab source code 62 D UNIX shell scripts for automatic measurement 73 3

6 Acknowledgments First, i would like to express my gratitude to my advisor, Ian Marsh, for his help, support, lessons taught and the friendship environment. A special thanks to (my examinator) Professor Olof Hagsand for his valuable help. Special thanks also goes to Professor Gerald Maguire for his help and motivation. Many thanks to Professor Gunnar Karlsson for his suggestions and books, and to all people from the Laboratory of Communication Networks (LCN) and SICS. Finally, i a very special thanks goes to my thesis colleague, Juan Carlos, for all his help and motivation. To Iara, Mama, Dady and Dada for their endless support and love, not only during this thesis but in my all life, To my dear girlfriend, Iarah, 4

7 Chapter 1 Introduction Voice over IP (VoIP) is already a reality in business, institutional and home networks. It has proved to be a serious competitor to the public switched telephone network, in terms of cost, eciency, quality, versatility and reliability. It started with small applications for private consumers to make calls from computer to computer. Then it progressed from computer to regular phone networks via gateways, over time it has gained acceptance amongst the business community as it could provide a whole telephony system over the already available IP network for a fraction of the cost of traditional telephony. As a result, VoIP provides the solution for the long time desired convergence between voice and data networks. However, in spite of its many advantages there are some issues. Classic telephony networks are designed to oer deterministic quality voice communication by using circuit-switched technology by using dedicated channels for each voice session. VoIP, on the other hand, as it is developed to work over IP networks, shares the drawbacks of packet-switched technology: there is no guaranteed quality of service. IP networks were not designed for voice, but for data, and they are based on the 'best eort' principle which means that packets are not guaranteed to be delivered to the receiving station. Therefore the higher level protocols are responsible for the reliable transmission of data. For data with real-time demands, the reliability mechanisms are simply too costly in terms of delay. Additionally, an IP network is a shared resource utilized by dierent applications and devices that compete for access to the channel. This competition can lead to bottlenecks, delays or even lost packets which is detrimental to real-time applications. In order for VoIP to work adequately, some requirements have to be fullled, or, in other words there are demands on the Quality of Service (QoS) from the network. The voice stream must not suer a delay higher than 150 ms, including processing delays added at the end systems plus the network's latency as this would lessen the interactivity of the conversation[rg03]. Furthermore, depending on the codec utilised to transform the analog voice signal into a digital stream of packets, a percentage of lost packets must be kept under a 5

8 certain minimum; if this is not controlled it may be impossible to reconstruct the voice at the listener in a comprehensible manner. There are mechanisms and technologies that provide the QoS required by VoIP for specic networks, but many do not support them [BBC + 98]. Not only has the technology of packet-switched networks matured, thus decreasing the dierence in terms of suitability for voice transmission between circuit-switched and packet-switched networks, but improvements in voice processing and concealment techniques make voice communication robust under poor network conditions. However, as the capacity of the networks increases, so does the number of users and the use of the network (P2P, backups, etc.) new applications tend to be more greedy in terms of their bandwidth usage. This generates both impairments to the QoS trac and the need for monitoring the network conditions to verify if real-time applications can operate with sucient quality. This monitoring, in the case of voice communication, it is not an easy task. The quality of a voice call not only depends on network conditions, but also on perceptual characteristics of the end users and how these network impairments aect the user's perception. Other factors are the user's expectations, since mood and human memory can increase or decrease the perceived quality. Furthermore, two identical sets of errors in the transmission can have dierent impact on the quality depending on the part of the speech in which they occur [LGR03]. This project was developed in the context of research programs conducted at the Swedish Institute of Computer Science (SICS) and at the Royal Institute of Technology, at the Laboratory for Communication Networks (LCN) in the Institute of Microelectronics and Information Technology (IMIT). 1.1 Motivation and goals Since b is by far the most popularly used wireless lan (WLAN) standard, the study of voice over these networks brings the greatest value to address the VoIP QoS issues. This includes the existing commercial and non-commercial networks. Therefore the motivation for measuring voice quality is an important activity for assessing its suitability as well as its limitations in real operating environments. Our concrete goal is to analyse the ability of b WLAN's to fulll the real-time the requirements of voice over WLANs. The objectives for our project, in increasing order of importance, are: Assess the suitability of VoIP quality of service in wireless LAN's considering the environment Merge the results with another masters student looking at the inuence of b MAC protocol layer inuence on VoIP quality Collect measurements into a repository for further study 6

9 1.2 Thesis structure In Chapter 2 the background material necessary to understand this thesis is presented. Chapter 3 outlines the approach we have taken as well as the method used to follow our chosen approach. Chapter 4 gives the results we obtained and forms the main contribution of this work. The results Section is followed by published works by researchers within the same area of VoIP measurements as well as a comparison of their results with ours in Chapter 5. In work such as this where there are many parameters and options available, we have included a discussion that gives the advantages and disadvantages of the approach and results we have taken. Clearly there is not a single (or even best) way to assess the quality of VoIP that utilizes wireless networks and we elude to this in Section 6. Chapter 7 contains the conclusions, again this is not a simple task but we attempt to draw together the big picture for VoIP in wireless scenarios. We have included some pointers for future study in Chapter 8 as well as the lessons we learned during this master thesis work in Chapter 9. The rst appendix contains a glossary of the acronyms used in this thesis. There are three additional appendices that give pure technical information for people who want recreate the measurements we have done. 7

10 Chapter 2 Background In this Chapter we present the most relevant concepts that need to be understood in order to understand the subsequent Chapters. 2.1 Network layers and protocols The b protocol stack is a simplied version of the OSI model and consists of ve layers: application, transport, network, data link and physical layer. The application, transport and network layers are usually implemented in software, while the data link and physical layers are commonly implemented in network interfaces (i.e. hardware). Each layer has its own protocols and the process of passing information between adjacent layers is generally dened by standard bodies. In Figure 2.1 we illustrate how data units are encapsulated and show the extra information needed by this layered approach as overhead. Overhead has a signicant impact on voice-over-wlan QoS since it increases the delay. We will now describe some of the layers shown in the Figure, particularly those relevant for VoIP communication. Figure 2.1: Network layers 8

11 Figure 2.2: MAC layer structure UDP protocol The transport layer generally provides services to the application layer using two protocols, TCP and UDP. TCP oers reliable packet delivery using control mechanisms; while UDP provides an unreliable, connectionless service and adds only an 8 byte header overhead. Because there is no mechanism to prevent packet losses, applications running over UDP are usually loss-tolerant IP protocol The Internet Protocol is the most used protocol at the network layer, and it is responsible for routing datagrams between hosts and routers. In this work we used IPv4 in the end systems that adds 20 bytes of overhead IEEE b standard data link and physical layers The data link layer is used to move datagrams from one node to the next node along a route, whilst the physical layer operates at the bit level eectively performing the same job. The important aspect of the MAC layer within this work is that it is able to retransmit data frames that were not acknowledged by the receiver. This reliability is important for the quality of VoIP sessions in poor quality situations, but adds delay to the communication. IEEE b standard splits the data link layer into Logic Link Control (LCC) and MAC sublayers. The physical layer is split into the Physical Layer Convergence Procedure (PLCP) and Physical Medium Dependent (PMD) sublayers. The standard species the functions of the MAC and both physical sub-layers, whilst the LCC sub layer is dened in the IEEE standard and the bridging between LANs is dened in IEEE standard. The IEEE b MAC layer provides two modes of accessing the medium, namely the Distributed Coordination Function (DCF) and the Point Coordination Function (PCF). DCF mode is based on the Carrier Sense Multiple Access with Collision Avoidance protocol (CSMA/CA) and we used this mode in our 9

12 experiments. PCF is out of the scope of this project since it is not supported by most network devices. The standard also provides two architectures with which to build wireless networks. The infra-structure mode, based on access points that mediating the communication between mobile stations within the same cell makes it possible to extend the networks, thus interconnecting several access points and wireless nodes can roam between the dierent access points within the same network. In the ad-hoc mode when there is no access point the stations communicate directly. 2.2 Real-time protocols RTP The RTP standard denes a packet structure for use in real-time applications, amongst others it includes elds for timestamps and sequence numbers necessary for synchronization. RTP runs over UDP however technically the RTP protocol is implemented in the application layer, however it is often viewed as a sublayer of the transport layer. Data encapsulation is performed at the end systems and the protocol does not provide any mechanism to prevent out of order packets or quality mechanisms. In order to provide control functions RTP is usually complemented with the RTCP protocol, described in the next Section RTCP The RTP control protocol, RTCP, is usually implemented in combination with the RTP protocol and is based on the periodic transmission of control packets. RTCP doesn't send a payload with application data, instead it sends statistical information that is necessary for the application program to provide feedback on the VoIP quality. The way this information is used by the application layer is not dened in the RFC3550 standard, and depends strictly on the application. RTCP and RTP packets are distinguished by using dierent port numbers typically the RTCP port is one higher than the RTP port, RTCP also uses UDP as the transport protocol. The sender report (SR) and receiver reports (RR) provide feedback about the reception quality. The only dierence between the two is that the sender report is used by active senders only and it has an extra eld with sender information. If a node sent no data since the last report, then the application sends a RR, otherwise it sends a SR. Source description items (SDES): SDES is a tree level structured header and contains information describing source parameters, such as the CNAME. BYE indicates the end of participation and that a source is no longer active. Application specic functions (APS) are to be used by newly developed applications and features, it is an experimental header. 10

13 Sender Application Receiver Application Microphone Loudspeaker A/D OS IP IP IP OS IP D/A LL LL LL LL End to end measurements Figure 2.3: Measurement delay 2.3 Voice QoS Good quality of service means providing satisfactory experience to the end users. More specically it usually entails holding error rates low and minimizing latency, loss and jitter during VoIP calls in our case. There are two classes of voice quality of service metrics: one is an objective metric which can be computed with high precision, and the other is a subjective metric involving humans listening and assigning a rating to the speech delity. Voice quality is related to three major factors: delay, loss, and jitter. Whilst data trac is mostly aected by packet loss and is more resilient to delay, voice calls are loss tolerant but more sensitive to delay and jitter (delay variance) End-to-end delay In voice communication delay usually refers to the end-to-end delay, that is the time a packet takes to travel from the sender process to the receiver process. This is shown in Figure 2.3. End-to-end delay has a signicant eect on the perceived quality of IP telephony, and it can be the case that it is not the same in both directions (so called asymmetric links). We assume that the end-toend delay is the sum of the processing, queuing, transmission and propagation delays Packetisation delay: The time required by the nodes (access point, router, end hosts etc.) to process the information and redirect the voice packets. The packetisation delay is generally in the order of milliseconds Queuing delay: The delay experienced while the packet is waiting to be transmitted. It depends on both the network trac intensity and network design (link, equipment, structure etc.) it is typically in the order of microseconds to milliseconds. 11

14 Serialization delay: Time required to put one packets onto the wireless link. Typically in the order of microseconds to milliseconds. The serialization delay can often be minimized thus increasing the link throughput Propagation delay: The time needed to propagate the information via the wireless and/or wired links that may exist between the sender and the receiver. Typically in the order of microseconds. To analyse delay we used the ITU-T G114 standard denitions [RG]. The relation between the end-to-end delay and voice quality is shown in Table 2.1. End-to-end delay (ms) lower than 150 between 150 and 400 higher than 400 Voice quality Good Acceptable Poor Table 2.1: End-to-end delay impact on voice quality Jitter Jitter is the packet inter-arrival statistical variation introduced by the IP network links (best eort network). Dierent denitions of jitter exist, however in this report we will adopt the IETF denition of jitter. Hence we dene jitter as the mean deviation of the packet spacing at the receiver compared to the sender packet spacing for a pair of packets Loss VoIP in b WLAN's runs over the UDP protocol which does not oer reliable delivery of the voice packets. During periods of congestion or interference packets may be lost. The voice application has to deal with and tolerate packet losses and depends primarily on the encoding/decoding techniques. Some techniques allow as much as 50% loss rate. As well as the absolute loss rate we are also concerned with the distribution of the losses, correlated losses can give rise to a much lower perceived quality than sporadic losses. The voice call generator we selected for our experiments and implements an 8-bit PCM modulation which can deal with 1% losses without loss concealment and up to 10% using loss concealment techniques. We dened a threshold of 5% and our quality analysis regarding losses is as represented in Table 2.2. Percentage of losses lower than 5 % higher than 5 % Voice quality Good Poor Table 2.2: Packets loss impact on voice quality 12

15 2.3.4 Subjective metrics The Mean Opinion Score (MOS) is a scale which rates the relative quality of voice conversations as perceived by human users. It is the most widely used subjective quality metric. Because the tests are done by humans subjects it is useful to evaluate the network capability to support audio conversations, however is too expensive and time consuming for our use. The MOS ratings are shown in Table 2.3. Rating Description 5 Imperceptible errors 4 Perceptible but annoying 3 Slightly annoying 2 Annoying 1 Very Annoying Table 2.3: Average MOS rating Objective metrics MOS values can also be calculated using objective metrics such as the ITU algorithm, Perceptual Evaluation of Speech Quality [Uni01]. The metric is less time consuming and cheaper than the MOS metric. Essentially, PESQ methods estimate the speech quality based on a psycho-acoustic model of the human perception by comparing the degraded speech sample with its clean version in the perceptual domain. Works using this method include statistical methods to predict speech quality [MCPA01a, MCPA01b]. Implementing an objective metric is proposed as future work. 2.4 Sister project Since this project has close ties to Juan Carlos Martin's thesis work Measurements of parameters that aect voice quality over IEEE b networks [Mar04], we provide some explanation how these two masters works are related. Their denition was done simultaneously in order to look at the quality of VoIP from both the application and medium access layer perspectives. Since we are dealing with wireless communication this is intuitively the correct approach to take in order to quantify and qualify how the two layers interact. This masters thesis concentrates on general measurements of VoIP quality using b networks in a variety of environmental situations and operating modes. The design of the experiments is more important in this thesis to capture the signicant factors that inuence the quality, i.e. to cover the common cases. The sister project is more focused on explaining why certain situations occurred (usually poor quality ones) by showing when the MAC protocol needed to perform 13

16 retransmissions of the data frames. Also the sister project can give the contribution to the loss, delay and jitter of the applications frames was. These two projects have both common and dierent facets to their approach, results and conclusions, so it recommended to look at this and the other project together in order to obtain a fuller picture of real-time voice communication in b environments. 14

17 Chapter 3 Approach and method 3.1 Approach The approach of this work is a systematic one. Our goal of making general, but careful, statements about voice over IP quality for b networks led us to a stepwise experimental approach. Experiments began with quality measurements in open space, theoretically two voice communicating nodes with their only impediment to (good) quality being physical distance. We hoped to be able to discount interference and obstacles from this experimental setup. From these measurements we moved inside to a typical oce environment in order to compare the previous results with line of sight measurements in a closed environment. By including obstacles between the (still only) two communicating nodes more complexity was added to the setup. Then the results of these three scenarios were compared. Adding more complexity we looked at the eect of competing trac on the same b channel and its disturbance on the VoIP ows. Finally an access point was added to the measurements in order to create a full-functioning measurement setup built on the experience (and results) of simpler systems studied previously. We also look at a number of smaller adjunct interesting cases such as the eect of microwave ovens (since they contend for the free frequency space), dierent hardware congurations and the eect of the RTS/CTS MAC protocol. In all of the measurements we concern ourselves with loss, delay and jitter of the communication. Our results focus on two aspects, rst, the typical or representative behavior of the communication, this is the general picture we would like to present. Second, the more interesting cases or anomalies where further investigation or research is needed. In this masters thesis, we do not delve into these, but mention them where worthwhile. 15

18 3.2 Method To analyze the suitability of IEEE b [oee99] networks to carry voice trac we performed a series of experiments covering a range of factors aecting the voice quality, including both network and link quality aspects. To compute the end-to-end delay, loss and jitter at the end-points we used a Sphone real-time extension (see Section ) and a Matlab tool (Section ) to process the results. Shell scripts were developed to automate the measurement process wherever possible and the Ethereal analyzer (Section ) was used to track all trac in the channel in order to monitor the MAC behavior. Our measurements were designed in increasing order of complexity and include two main sets of experiments: ad-hoc and infra-structure experiments. Using an 80 second voice session representing business call length, the following set of experiments were used to assess the quality of VoIP in b networks: 1. Ad-hoc mode Line of sight in open space Line of sight in an oce Channel competition Obstacles in an oce 2. Infra-structure mode Competing trac Ad-hoc mode In ad-hoc mode we were interested in gaining an insight into the suitability of b networks to run applications requiring direct voice communications between two single nodes, as is the case of DECT phones in an oce environment. Also these measurements represent the simplest case for quality measurements. The basic conguration of the ad-hoc experiments is represented in Figure 3.1. A sender is attached to a packet capture machine using a wired link. The hardware and conguration details can be found in Appendix B. We started with measurements in free space to achieve some understanding of the relation between the link degradation and the distance. These results served as a reference to compare the following measurements with this 'base' result. As far as we could adjudge the spectrum was free from interfering transmissions, other base stations as well as machinery operating in the same 2.4 Ghz frequency range. The goal is to see if unexpected eects are anyway related to distance. For the next series of measurements we moved inside the Swedish Institute of Computer Science (SICS) in Kista, Sweden. It is representative of a typical oce network with a layout of the premises as shown in Figure 3.2 We proceeded with the line of sight measurements and then compared these results with the ones we obtained in free space. This allowed us to conclude 16

19 Figure 3.1: Ad-hoc measurement setup (receiver) Figure 3.2: SICS layout 17

20 Figure 3.3: Ad-hoc measurement setup about the eects of the closed environment on the link quality. Thereafter measurements with obstacles between the sender and receiver nodes were taken. With the sender situated in one oce and the receiver at dierent locations within the oce space, we were able to simulate communication within the premises. This is typical of communication of an employee communicating with a base station or a secretary forwarding a call that arrives to a central phone. We also performed experiments with intervening windows, walls and other obstacles plus one with the microwave oven switched on whilst communicating. The objective was to include a wide range of situations that are typically found and to study their eect on the voice data. The nal measurement in ad-hoc mode was to study the eect of competing trac in an ad-hoc network, as represented in Figure 3.3. An increasing number of TCP sessions between the nodes was generated and we analyzed the eects of this external load on the quality of VoIP sessions for a single b channel Infra-structure mode In these experiments we only address the problem of competing voice and TCP trac in the same network using the test-bed conguration as shown in Figure 3.4. First we used one node to generate a single TCP ow using a trac generator. We repeated the procedure for an increasing number of TCP ows so we could infer how it impacts on voice performance. 18

21 Figure 3.4: Infra-structure measurement setup Tools Sphone We have used Sphone, a low-delay VoIP communication tool developed at IMIT/KTH, to generate the IP-telephony sessions [OHH03]. The aim of Sphone is to provide a platform for Internet telephony and it is distributed under the terms of the GNU General Public License as published by the Free Software Foundation. Sphone was chosen because it provides an implementation of the RTP and RTCP protocols which we could modify, plus it has been used in measurement activities before [Li02]. Sphone is implemented in (ANSI) C and it implements two basic modules: a sender and a receiver Extending Sphone: the real-time tool We extended Sphone to provide users with real-time QoS feedback and to generate more comprehensive log-les for post processing purposes. With the realtime tool we compute IP quality metrics and display these values at congurable intervals so the user can have a clear perception of the link quality in real-time. For each session a corresponding log le is created and stored as described in Section It is important to note that Sphone was designed for research purposes and the IP metrics and log les are useful for this community. These metrics also provide the basis to develop an objective link quality estimation for future use in both Sphone and other tools. To keep the code modularized we added all our functions in a new le named qos.c and the corresponding qos.h. 19

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