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1 Managing VoIP Quality Expert Lesson Voice traffic presents many challenges for the network manager. In addition to keeping packets flowing over the LAN and WAN, with voice or IP telephony, the network manager must be concerned with application performance management, real-time issues such as delay and jitter, as well as the end user's experience. This expert guide will explore the issues that can impact your network when VoIP comes into play, and our expert will discuss strategies for keeping your voice and data traffic running smoothly on one network before end users start to complain about call quality. Sponsored By:

2 Table of Contents Managing VoIP Quality E-Guide Table of Contents: QoS Is a Must for VoIP Quality Four Critical VoIP QoS Components User Experience Measurement Is Key to VoIP Quality Related Resources from CA Sponsored by: Page 2 of 12

3 QoS Is a Must for VoIP Quality QoS Is a Must for VoIP Quality For those of us with a data-centric worldview, it is tempting to dismiss voice/data convergence as just one more application joining the data network bit stream. But VoIP is not just another application merging with others like it it is a different beast. And the sooner you understand and prepare for that reality, the better off you and your network users will be. One difference is a result not of the application itself but of users expectations. The plain old telephone service (POTS) world has set the expectation that voice calls will consistently be free from defects like echo, noise and delay and that the dial tone will always be there. Except during emergencies or on Mother s Day, you expect to get through and if you can t, you get annoyed. Other differences are technical. Real-time traffic is just not the same as data traffic. Voice streams differ profoundly from their data counterparts, and they need special treatment because packet loss, delay and jitter cause bad voice call quality. The most pertinent technical difference between data and voice is that voice (and video) uses UDP instead of TCP. To explore what that means for voice performance, let s look at how the two protocols handle packet loss. Network managers often tell us that they should not have any problem with packet loss because their links are lightly utilized. The truth is that even very lightly utilized links can and will experience packet loss. Here s why. Data applications move blocks of information from one computer to another, which leads to the bursty bandwidth usage shown in Figure 1, and to packet loss. TCP is designed to overcome packet loss easily, and as long as the loss is not excessive, TCP works well. In fact, TCP often creates packet loss as a way to find out how much bandwidth is available. Remember that TCP is an end-to-end protocol, executed by the sending and receiving computers. The network speed at either end of the link may be very high (100 Mbps or 1 Gbps), but there may also be a slow-speed link along the path. TCP starts off slowly and increases its speed until packet loss occurs. Then it backs down to half that speed and starts to creep up again. This is how it determines available bandwidth and how it dynamically shares that bandwidth with other TCP flows. Sponsored by: Page 3 of 12

4 QoS Is a Must for VoIP Quality Figure 1. Typical data traffic Unlike data traffic, voice traffic moves as a continuous stream of data that recreates the analog event of human conversation. Bandwidth use is constant, as Figure 2 shows. Voice traffic streams use UDP, which unlike TCP does not recover lost packets and degrades quickly when packets are not delivered. UDP is what we call a send and pray protocol like the post office. We put packets (or letters) into the network and just hope they get there. The UDP stream doesn t know anything about available bandwidth and makes no adjustment if bandwidth is insufficient. When bandwidth is constrained, packets are lost. Because UDP does not recover those packets, the receiver does not get all the data, and voice quality dives. Figure 2. Typical voice traffic Sponsored by: Page 4 of 12

5 QoS Is a Must for VoIP Quality But, you ask, why should there be any packet loss if utilization is low say in the 30% range? Because utilization measurements are averaged over some time period often minutes and peak utilization occurs in bursts that are often very short, such as during Web page loading. During those short bursts, TCP tries to use all the available bandwidth and causes packet loss. It s even common to see packet loss on 10 Gb backbone links that are 3% utilized because of such TCP peaks. Figure 3. QoS keeps data bursts from overrunning voice To overcome this problem, you need QoS to make sure your voice streams get through, even during peak bursts. Figure 3 shows how QoS gives priority to voice traffic, which holds down the data packet peaks and prevents packet loss and jitter in the voice streams. This capability is not just a nice-to-have, it s a musthave if you want your VoIP users to remain productive and satisfied. In our next section, we give some tips on how to plan and implement a successful VoIP QoS strategy. Sponsored by: Page 5 of 12

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7 Four Critical VoIP QoS Components Four Critical VoIP QoS Components Implementing QoS involves more than just turning on a router feature and unfortunately there are many ways to skin the QoS cat, so you must make time to plan and design a QoS deployment that fits your unique needs. There are four critical components of a QoS deployment, and if you fail to implement all of them, your solution will not deliver its full potential. Tip 1: Classify your voice traffic For starters, you need to choose the best scheme to mark packets for priority treatment. Classification is the job of determining which streams get high priority on the way to their destination and which do not. Of course, every user would like his or her applications to run with the highest priority but the outcome would be the same as doing nothing. QoS is a zero-sum game requiring decisions about who gets waved through and who waits. Because it is real time and has fragile performance characteristics, VoIP needs to receive the highest-priority classification. We recommend using mechanisms that enable the network to verify endpoints. If the network can verify that the device at a specific IP address is an IP phone from a vendor of choice, then the router can trust the traffic priority markings that phone provides. In this way, the network protects itself from inappropriate use and the endpoint gets to determine which packets receive high priority and which receive only besteffort support. Before you begin marking packets, however, NetForecast recommends that you formulate a master plan for what traffic will use which service levels, how many service levels you will have, and what markings to apply to each traffic class. You also need to map your priority classes across the layer 2/layer 3 boundary and map your priority classes to your carrier s service offerings. Tip 2: Enable class-of-service mechanisms Now it s time to turn on the mechanisms in your network routers and switches that will implement the priority treatment for VoIP. The predominant class-of-service (CoS) mechanisms are DiffServ for routed infrastructure, and IEEE 802.1p for switched infrastructure. These mechanisms are embedded into most network infrastructure and are just waiting to be turned on if they are not already. Your network team can enable CoS through the appropriate commands in the router and switch configuration files. You will find that turning them on is easy. Tip 3: Plan and manage your bandwidth Our third tip for ensuring successful VoIP quality is to understand, plan for and manage your bandwidth. Upon close examination, you will find that DiffServ and IEEE 802.1p are class-of-service not full-fledged quality-of-service mechanisms. This means they provide different levels of service, but they do not control oversubscription. For that, you need additional mechanisms to ensure that you do not max out the classes of service you establish in the network because if you do, voice quality will suffer. Sponsored by: Page 7 of 12

8 Four Critical VoIP QoS Components To deliver solid VoIP service quality, you must first estimate the number of calls you expect to occur simultaneously. Then you need to assess whether available bandwidth will support projected demand or whether you need link upgrades. At this point, financial reality will kick in, because chances are that your budget will not support all the bandwidth you need at all locations. You will therefore need to manage the number of users who can access the bandwidth at once, and you will need the same type of feedback that the public switched telephone network (PSTN) gives when there is insufficient bandwidth for a call the VoIP equivalent of a trunk busy signal. IP-PBXs have this function built in. The IP-PBX is programmed to know about the network topology and is then given parameters for how many voice calls can occur simultaneously between location A and location B. When this limit is reached, the IP-PBX will either give the next caller a busy signal or route the call to an alternate path (such as the PSTN). Tip 4: Monitor VoIP performance Our final tip for successful VoIP quality is to monitor performance. You need to know whether your network is providing high-quality transport for voice streams and high-quality voice reproduction. If there are issues, you need to know about them before your users have a frustrating experience. Because voice has special needs, new types of monitoring tools are required for this task. In our next section, we give an overview of those tools. Sponsored by: Page 8 of 12

9 User Experience Measurement Is Key to VoIP Quality User Experience Measurement Is Key to VoIP Quality Measuring the user experience is essential to achieve consistently good VoIP quality. Why? Because we re talking about talking, not ! A VoIP application s raison d être is to recreate a real-world, ongoing event a conversation between living, breathing people who have high expectations for a quality experience. For a good VoIP experience, the data necessary to reconstruct the voice needs to be delivered quickly and consistently to maintain the sense that the interlocutors are in the same room and communications are instantaneous. Before we dive into how to measure the user experience, let s clarify the difference between quality of experience (QoE) and quality of service (QoS). QoS refers to how well the network delivers packets. QoS techniques like classification and DiffServ differentiate packet streams in the network and enhance the delivery of streams that are sensitive to network impairments like packet loss and jitter. QoE, on the other hand, involves measuring and understanding the actual user experience. Here is the challenge that calls for QoS measurement tools. Unlike data, real-time voice streams are carried by UDP not TCP, and UDP does not recover lost packets as TCP does. For TCP-based applications, a little packet loss causes a minor application responsiveness slowdown; but for voice, a little packet loss dramatically degrades the user experience. Traditional data networking test tools are not designed to ferret out problems that cause packet loss in real-time streams. The only way to know whether a network path is really losing packets is to test the actual stream or to simulate that stream with equivalent data. The measurements must take into account the whole network from end to end. We can hear an indignant network engineer reading this and saying: My tools are monitoring packet loss, so I am all set. But are you? If you think you are, it probably means you have a tool that monitors for packet drops on key network routers. The tool probably looks for output queue drops on critical links where large numbers of links converge, for example, or boundaries where traffic flows from a high-speed link (e.g., a LAN link) to a lower-speed link (e.g., a WAN link). This is good information, but it is not enough. The fact is that packet loss can occur anywhere along the path between endpoints. A common cause of loss is a half/full duplex mismatch that causes a layer 2 error which is not detected by the Ethernet collision mechanism because of the mismatch. The packet is dropped because it is incomplete or its checksum is bad, so no router queue ever sees the packet. The router reports no packet drops, but this packet never even got there! The same can be true for a noisy copper line, bad Ethernet cables or even a router interface that is unmonitored. But the application is still missing data, and the voice quality will degrade. But just delivering packets to the far end does not guarantee a quality experience. Suppose, for example, that the voice signal reproduced by the VoIP system has a poor signal-to-noise ratio. Even if every packet is delivered, noise can make voice unintelligible. Sponsored by: Page 9 of 12

10 User Experience Measurement Is Key to VoIP Quality In a typical VoIP deployment, a voice stream may start in the PSTN, pass through a gateway into a VoIP environment, be carried across a SIP trunk to a remote site, have its compression type recoded for compatibility with a second VoIP vendor s equipment, and finally be reconstructed on a VoIP handset. More transitions than this are both possible and probable. A number of problems can be introduced, including noise, poor recoding, echo caused by a long end-to-end delay, and quality loss resulting from the type of compression used. The resulting voice quality experienced may be bad, even though the network delivers 100% of the packets on time. You need quality of experience (QoE) measurement tools to get to the bottom of all these problems. QoE tools reconstruct the voice signal in a measurement tool and run signal-processing algorithms to assess call quality. QoE measurement tools generally deliver a score that more accurately reflects the true user experience than simply watching network packet characteristics. QoE standards have been defined by the ITU to provide a consistent way of measuring the expected quality of a voice call. This is very useful for comparing measurements among vendors, service providers, and different implementations. Although new to market, QoE measurement tools are maturing and improving quickly. If you are planning or are in the midst of an important VoIP deployment, we suggest you procure one of these tools to help you establish and ensure ongoing VoIP quality. About the authors: John Bartlett is principal of NetForecast. He is a leading authority on real-time traffic, Internet performance and Quality of Service (QoS) techniques. NetForecast specializes in network-based application performance issues for transaction-based and real-time applications. John has 28 years of experience in the semiconductor, computer and communications fields in marketing, sales, engineering, manufacturing and consulting roles. He has contributed to microprocessor, computer and network equipment design for over 40 products. He has been consulting since Prior to working as a consultant, John was a founder and VP of Engineering and Manufacturing at Agile Networks, now part of Lucent Technologies. A popular speaker at VoiceCon and InterOp, John gives tutorials on Quality of Service, performance issues and real time traffic behavior on LANs and the Internet. Rebecca Wetzel is a principal of NetForecast. She is also president of the marketing consulting firm, Wetzel Consulting LLC, where she provides data communications industry insight and helps vendors and service providers develop and deploy successful marketing strategies. Ms. Wetzel draws on her direct experience of more than 20 years in data networking. Her areas of expertise include application performance measurement, control and acceleration technologies, network and application security, voice over IP, virtual private networking solutions, as well as a variety of other emerging technologies such as anti-phishing and spam mitigation. In addition, for many years she has Sponsored by: Page 10 of 12

11 User Experience Measurement Is Key to VoIP Quality performed customer-focused market research to help Internet services providers rate their performance relative to competitors, and determine what new service offerings their customers want. Co-author of the book Internet Telephony for Dummies, Rebecca regularly contributes to and is quoted by such publications as the Wall Street Journal, Network World, and Business Communications Review. She has also authored dozens of white papers and reports. Sponsored by: Page 11 of 12

12 Related Resources from CA Related Resources from CA Ensuring Successful Voice Over IP Deployment Network and Voice Management for Evolving Business Environments The Definitive Guide to Converged Network Management Network Change and Configuration Management: Optimize Reliability, Minimize Risks and Reduce Costs Incomparison Network and Voice Management Software About CA CA is one of the world's largest IT management software providers. Our software and expertise unify and simplify complex IT environments in a secure way across the enterprise for greater business results. More than 3000 developers worldwide create and deliver IT management software that keeps our vision real. CA is a global company with headquarters in the US and 150 offices in more than 45 countries. We serve more than 98% of Fortune 1000 companies, as well as government entities, educational institutions and thousands of other companies in diverse industries worldwide. Driving our next level of growth is a our four-part strategy of product development, leveraging partners, global expansion and strategic acquisitions all with the goal of helping our customers realize the full power of IT to drive their business. Sponsored by: Page 12 of 12