Maximizing Voice over Internet Protocol (VoIP) Networks

TechRepublicRealWorld Guide

Maximizing Voice over Internet Protocol(VoIP) Networks

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Table of Contents

Integrate VoIP with your existing network:Planning considerations ........................................................................3Perform effective VoIP network capacity planning ..........................6Successfully deploy and maintain VoIP with these best practices............................................................................................9Improve VoIP management with these best practices ....................12

TechRepublic Real World Guide: Maximizing Voice over Internet Protocol (VoIP) Networks

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TechRepublic Real World Guide:Enterprise IP Phone Systems

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As organizations increasingly recognize and require the benefits voice overInternet Protocol (VoIP) offers, they stop asking “Why?” and start asking“How?”. A successful VoIP implementation requires a keen understanding of

how voice services will integrate with the existing network. Organizations must knowhow voice and data traffic will coexist, what infrastructure additions will be requiredand whether major network re-designs will be necessary. This document takes a macro-scopic look at how VoIP technology integrates with existing network infrastructure.

Network infrastructureThe first and foremost consideration with any VoIP implementation is the cumulativenetwork infrastructure (switches, routers, firewalls and the like) to which you will add VoIP. If you currently run an IP network, or even a hybrid IP network, VoIP tech-nology lets you pass voice traffic, encapsulated in IP packets, through your existing network hardware. While the utilization of existing infrastructure is a key VoIP benefit,you must allow for your current network’s capabilities. You should know whether yourswitches can place voice and data networks into separate VLANs. You should alsounderstand the Quality of Service (QoS) capabilities your switches and routers offer.

Along with your network hardware’s supported features, a successful VoIP integra-tion will depend on the hardware’s physical capacity. Adding an additional traffic typeto you existing network will increase network resource consumption. Bandwidthrequirements will rise and with them memory and CPU utilization. Minimum hardwarerequirements should be followed to ensure successful delivery of both voice and datatraffic. For example, a 100Mb switched connection to the desktop is a commonrequirement. When your voice conversation depends on the underlying network’s avail-ability, you must ensure that traffic can be fully supported; otherwise, the voice com-munications quality will deteriorate as dropped packets cause choppy audio and speechdelays. Data communications are resilient to delayed or lost packets, voice is not.

Before beginning a voice and data integration project, you should have accuratebaseline data for the existing network’s performance. Evaluate possible bottlenecks,network errors, and average bandwidth usage before putting phones on the network.This should be done before you begin evaluating a VoIP solution. Reliable baselinedata will help you choose the VoIP implementation that provides the most bang foryour buck.

Legacy telephony equipmentUnless you are deploying VoIP in a completely new environment, your VoIP solutionwill likely need to coexist, at least for a time, with your existing PBX or key switch telephone system. VoIP and traditional phones can coexist separately, allowing for a“flash” cutover—where you move from the old system to VoIP on a scheduled date.



Integrate VoIP with your existingnetwork: Planning considerationsBy Thomas Nooning, CCNA, CCDA

VoIP Talking PointsIP and hybrid IP networkscan use VoIP technologyVoIP traffic will increasebandwidth and resourceusageSet minimum hardwarerequirements to ensure QoSAnalyze baseline network data before VoIP integrationLegacy telephone and VoIP systems can coexistBoth flash cutovers andstaggered migrations arepossibleNot all VoIP solutions support every voicemailmessaging protocolAdditional PSTN lines will be needed tosimultaneously run VoIPand legacy systems

If you prefer a staggered migration, you can run both systems simultaneously. While astaggered approach lets you test the VoIP waters before taking the plunge, it doesrequire that you link your VoIP solution with your existing phone system.

How you link your VoIP solution and your existing phone system will depend oneach system’s specific characteristics, but you can smooth the process with a few com-mon migration techniques Many companies use a dial plan with non-overlappingphone numbers to simplify call routing. If your existing system uses 5XXX and yourVoIP system uses 3XXX, both system will know how to route specific calls. You canalso connect geographically distant legacy phone systems with VoIP technology over aWAN link. This allows you to test a VoIP solution and save on long distance costs.Some VoIP systems are merely “IP-enabled” PBX’s that natively perform this function.

A variety of PBX’s and key systems exist, but almost any two systems can be madeto communication with each other. You typically create the link by emulating either astation-side or a trunk-side connection. Either system can then translate the phonenumbers to achieve interoperability. With large-scale VoIP rollouts this can be a criticalprocess. It is also important to verify the signaling types supported by both the legacyand VoIP systems.

Legacy phone system often include both a standalone voicemail system and a PBX.You should verify that your new VoIP solution will support your existing voicemail sys-tem. Voicemail systems use specific messaging protocols and not every VoIP systemsupports all the available protocols. Some VoIP solutions may require a new voicemailsystem and this condition may sway your choice of VoIP provider.

Public Switched Telephone Network (PSTN) connectivityEven if your entire enterprise is going completely VoIP, at some point you will need toreach the outside world. Small sites may only have a few business lines. Larger organi-zations may have one or many digital Primary Rate Interface (PRI) circuits. Unless youplan to change your outbound connection, you should choose a VoIP solution thatworks with your existing telephone circuit. VoIP equipment is designed to work withcommonly available telephone circuit types. You should search for an ideal solutionthat allows you simply unplug your exiting circuit from the PBX and connect it directlyto the VoIP gateway. To ensure interoperability you should also confirm compatibilityof signaling types, framing, and line coding.

In you plan to concurrently operate your legacy and VoIP systems, you will likelyneed separate Public Switched Telephone Network (PSTN) lines for each. It is unlikelyboth systems would be able to share the same local carrier lines. If you use a PRI cir-cuit and are using Direct Inward Dial (DID), you can port the new phone number tothe new PRI as you go. The approach allows some users to use the new system withtheir existing phone numbers and other users to operate on the old system as before.

The bottom lineIntegrating VoIP with your existing network takes a significant amount of careful plan-ning. You must ensure the existing network infrastructure can support the additionalbandwidth and performance requirements. Ideally, the equipment should be able tologically separate voice and data networks, applying specific levels of service to each.



Concurrently running your VoIP solution with your legacy phone system is a goodway to test the VoIP waters. This staggered approach will require linking the two sys-tems and you should verify interoperability of PBX or key systems and voicemail priorto deployment. It is highly likely that you will also require additional PSTN connectionswhile you simultaneously operate the two networks.

While seamlessly integrating VoIP with your existing network can be a complexprocess, careful planning will help you reach the end goal of a IP-based telephony net-work that provides additional features and cost savings.



Network engineers, analysts and administrators characterize network capacity as the amount of traffic the network is designed to handle. When discussingvoice over Internet Protocol (VoIP), network capacity becomes more a

measure of how many simultaneous calls the network can process. This concept of“peak load”, the maximum assumed volume that the network should be able to handle,will be the basis of your VoIP capacity planning. During your VoIP capacity planningprocess you should consider the following:

Your local area network (LAN) and/or wide area network (WAN) designExisting data traffic on the networkThe voice codecs your VoIP solution will useConnectivity to the Public Switched Telephone Network (PSTN)Your network’s hardware infrastructureVoIP and network redundancy

The document will identify and explain the key steps in VoIP network capacity plan-ning.

Mapping the current network and gathering baselinenetwork dataBegin your VoIP capacity planning process by developing a clear picture of your exist-ing network infrastructure. Identify each network link that will transmit voice trafficand document the link’s bandwidth. This information will help you recognize and over-come potential bottlenecks. Without knowing your current network utilization, youcan’t accurately forecast how adding voice traffic will impact that utilization.

Whether a bandwidth problem impacts your VoIP implementation or not dependslargely on your existing network’s layout. Network capacity is unlikely to block a VoIPdeployment on a single-site network with high-speed infrastructure. Common network-ing problems, such as duplex settings and broadcast storms, will impact overall networkbandwidth, but most modern LANs can easily pass a high number of VoIP calls.Bandwidth can be a roadblock when deploying VoIP over a WAN with multiple loca-tions. WAN links are largely serial-based connections operating on T1 or fractional T1lines. These are bandwidth bottlenecks on many networks and will impact VoIP com-munications.

Once you accurately document your existing network’s maximum capacity, you’llthen need to determine the bandwidth you’re network applications are currently using.A remote site connected via a 768Kbps leased line can easily support 10 VoIP phones,but if that link is oversubscribed, it many not be able to handle both voice and datatraffic. For example, standard VoIP quality of service (QoS) guidelines specify that thenetwork transmit voice traffic ahead of data traffic. This improves voice service, but



Perform effective VoIP networkcapacity planning By Thomas Nooning, CCNA, CCDA

VoIP Capacity PlanningTalking Points

Map your existing networkinfrastructureIdentify each network link’sbandwidth and potential bottlenecksGather baseline networktraffic dataWAN links can be VoIPtraffic bottlenecksVoIP QoS in a tight bandwidth environment can negatively impact data trafficMeasure capacity needs on average and peak usage dataRemote office usage maywarrant increasing thespeed of existing circuits or adding new circuitsSimulate VoIP traffic to collect baseline VoIP dataRemote locations may need local PSTN gatewaysEnsure the network hardware can handle theincrease in traffic after VoIP deployment

can negatively impact other network applications. Large file transfers that once tookfive minutes, may take 20 minutes or more during heavy call periods.

Using standard PRI or voice T1 lines, you will likely consume all available channelsand be unable to place additional VoIP calls. Users will receive a fast-busy or an opera-tor error. If enough bandwidth isn’t available, heavy VoIP traffic can cause data circuitto constantly send data at their maximum bandwidth and eventually begin droppingpackets. Even with QoS measures and compression in place, having more VoIP callsthan the network can handle will also negatively impact call quality. Some vendors sup-port admission controls that monitor bandwidth usage and generate an alert end userswhen traffic is too heavy to place a call. Needless to say, waiting for an ebb in networktraffic is a poor user experience.

To determine each connection’s average and peak usage you can usually pull callrecords from your existing PBX and/or request historical call records from your tele-phone company. You average and peak usage data will help you determine how muchPSTN connectivity you should retain and how much bandwidth you may need to add.

With you network mapping and baseline data in hand, you should carefully considerincreasing the speed of existing circuits or adding additional lines to remote locations.You should base this decision on each location’s number of users and peak usage. Asmetropolitan-area network (MAN) technologies, like Metro Ethernet, proliferate,adding the higher bandwidth lines these services allow may cost less than adding moretraditional circuits.

Performing VoIP baselinesOnce you have a general VoIP traffic forecast, you can begin to test how the forecastedtraffic will actually impact your network. Standard bandwidth monitoring techniquesand protocol analyzers can provide good, basic information on much bandwidth you’reusing and help identify potential problems. You can perform a more detailed analysisusing a variety of VoIP-specific tools. These monitoring tools will accurately replicatedifferent types of voice traffic and watch for errors. They can monitor for VoIP prob-lems such as jitter and delay. Jitter is the measurement of transit delay in voice packetsand can be caused by many reasons. The end result is lower quality communications.You can select different VoIP codecs and simulate different call loads during real-timetests. Different codecs, such as G.711 and G.729, provide different sampling rates thataffect packet size. Selecting a lower quality codec can dramatically affect the bandwidthusage on network links. During your test you should also test compression, which canincrease available network resources and allow for more simultaneous phone calls.

PSTN ConnectivityDuring your VoIP capacity planning, you’ll need to determine if remote locations willcontinue to have PSTN connectivity or if that connectivity will be consolidated at acentral location. All phone calls would pass through WAN links to a central locationwhere the calls would proceed out the associated PSTN gateway. Centralizing PSTNconnectivity can simplify VoIP network planning and consolidating hardware candecrease costs and increase redundancy. PSTN consolidation can also impact VoIP



network capacity. The remote location’s size and bandwidth requirements will deter-mine the impact’s magnitude.

When locations reach a specific end user number, your best option may be a localconnection to the PSTN for local and offnet, long distance calls. For example, if yourremote locations incur long distance charges to call each other or a home office, itmight be better to pass internal voice traffic through your WAN but pass local, outsidecalls through a local PSTN gateway. You can easily accomplish this by routing callsbased on destination area code. Again, your PBX records and telephone company his-tory can help you determine how much call traffic falls into each category.

Hardware considerationsHardware directly influences network capacity. During your VoIP capacity planningprocess, you should carefully evaluate your existing hardware infrastructure and futurehardware needs. Adding VoIP to your network may double overall traffic. You mustensure the network’s core infrastructure and distribution-level points can handle theincrease. You should establish 100Mbps, full duplex connections to every phone. Mostmodern Ethernet hardware can currently support this connection type. If your net-work hardware is modular and lets you easily increase port density, you will likely beable to use your existing equipment. If not, you may need to purchase new hardware.You must also guarantee that your uplinks to the network core can handle the addition-al traffic load. The PSTN gateway is another hardware consideration. Adding additionalvoice modules to you PSTN gateway may require an expensive chassis upgrade.

Your hardware will also influence your network’s capability for QoS, compression,and voice support. As these tools and techniques can increase network bandwidth, con-sider them as carefully as you would fat pipes and fast backbones. The ability to grow iskey, and whenever possible VoIP should be rolled out in stages and capacity evaluatedat each step.

The bottom linePlanning for VoIP capacity involves a number of different steps, from the bandwidth avail-able on LAN and WAN links, to PSTN connectivity and hardware choices. VoIP, while notexactly “plug and play”, is a very feasible option given proper planning. WAN links willalmost always be choke points and should be evaluated to make sure they can support thenecessary voice traffic. Connections to the PSTN should be intelligently placed where theycan be most beneficial. Finally, you should thoroughly examine the network hardware infra-structure to ensure support for both immediate and future VoIP capacity needs.



Your VoIP vendor choice will influence your deployment and maintenanceprocesses, but following general VoIP best practices will also increase the project’s success. You can separate and handle voice traffic through standard,

vendor-neutral protocols and design practices. Furthermore, general layout and designmodels for integrated VoIP and data networks provide a strong base for many VoIPdeployments. VLANs, QoS, codecs, and compression will each help you build yourVoIP solution atop your existing network. This document identifies and explains general best methods for deploying and maintaining successful VoIP networks.

Use standard network monitoring techniques and VoIP-specific tools to continuouslymonitor the network’s health and identify problems.

VoIP ModelsRegardless of your VoIP network’s specific design, your should follow the standardguidelines appropriate for the overall VoIP network model. This document examinestwo widespread VoIP network models –– a centralized design and a distributed design.Centralized models provide call setup and teardown through a VoIP core involving oneor more soft-PBXs. A distributed model performs call setup and teardown at multiplelocations. The location are normally arranged geographically with links for inter-siteconnectivity. The centralized model has the advantage of hardware and PSTN consoli-dation. The distributed model is best used with multiple sites of equivalent size wheremost traffic remains local.

Redundancy is a critical success factor for a centralized model and is usually accom-plished with two or more VoIP servers that can each handle the total load. Theseservers should be dual-homed with two NICs going to separate switches. Backuppower should be provided via a UPS with enough available load to outlast multiplebrownouts as well as multi-hour blackouts. The entire phone system depends on theseservers retaining connectivity.

While a centralized design will need to be highly resilient, it usually involves lesshardware overall. PSTN connectivity can be consolidated at one location with a localgateway or redundant gateways. Organizations typically use a centralized model whenthey have a large central site, such a corporate headquarters, and several, smaller remotelocations. WAN links connect sites into the main location and provide administrationof voice communications. Call setup and teardown will be handled over the WANlinks, but local site calling will remain local. The centralized model’s disadvantage is therequirement to maintain connectivity to the central site from each location. WAN linkredundancy is typically expensive, but you can mitigate the cost through local failoveroptions. Many vendors offer redundancy capability, usually through a smaller-endPSTN gateway and minimal phone lines.



Successfully deploy and maintainVoIP with these best practicesBy Thomas Nooning, CCNA, CCDA

VoIP Deployment andMaintenance TalkingPoints

Choose a general VoIPnetwork design that best fits your organization.Centralized VoIP networkslet you consolidate hardware and PSTN connectivity.Centralized VoIP networksrequire a highly resilientcentral site.Distributed VoIP networksoften require more hardware than centralizednetworks.Distributed VoIP networksdon’t require as muchredundant connectivity ascentralized networks.QoS allows you to separatevoice and data traffic—treating each with distincttransmission policies.Compression reduces packet size and increasesavailable bandwidth.Codecs that provide highcall quality also requiremore bandwidth.The VoIP network shouldinclude redundant hardware,redundant links, and ade-quate power backups.

In a distributed model, multiple large sites usually require localized VoIP servers.These sites will pass inter-site voice calls through trunk lines that link the sites. Thebenefit here is a decreased need for redundant connectivity. The distributed model mayincrease however, to the overall hardware costs and design complexity.

Network considerationsDesign consideration will greatly influence the overall success when integrating VoIPwith an existing network. Network equipment will support virtual LANs (VLANs),quality of service (QoS), and compression. Using VLANs, you can separate voice anddata traffic but have them co-exists on the same medium. VLANs offer a simple wayto provide a distinct level of service as well allowing the individual traffic types to bemonitored.

QoS is a method of distinguishing IP packets so they can be treated by distinct policies. Your VoIP implementation should prioritize traffic to prevent existing datatraffic from undermining voice communication integrity. This process requires match-ing specific fields within either the Layer 2 or Layer 3 headers inside the data units.Layer 2 QoS, for example 802.1p, allows for multiple levels of prioritization by taggingframes with certain fields. At the switch port level, QoS can then be provided to trafficentering the port. Therefore, as data and voice traffic simultaneously enter the device,voice traffic can be sent first and the data traffic queued for delivery.

On Layer 3 links, such as WAN links between sites, QoS can also be used to matchon fields within the IP header. Again, this provides queues for data traffic for deliverywhile sending voice traffic with priority. As these QoS processes are fairly detailed,you must consider how much bandwidth you will allot to individual traffic types. Forexample, you could provide 75 percent of a WAN circuit to carry voice traffic, leave 5 percent for call setup and teardown activity, and reserve 20 percent exclusively fordata traffic. You may also want to identify highly-critical data traffic that you placesomewhere below voice, but above the general data queue. When dealing with limitedbandwidth, you should allow ample time to create an effective QoS policy. Your QoSpolicy will ultimately determine inter-site voice quality as otherwise the network sendstraffic on a first-come, first-served basis.

After QoS, compression and codes should be your next consideration. Compressionwill create smaller packets by shrinking duplicated information. Often seen on low-speed WAN links, compression increases the overall available bandwidth. Although,voice networks usually only compress packet headers, the practice will have a positiveimpact. The specific VoIP codec your solutions uses will also directly affect bandwidth.A high-quality codec like G.711 uses a sampling rate which translates in higher per-second bandwidth rates. G.711 will require approximately 90Kbps one-way for voicecommunication. G.729 on the other hand uses a lesser sampling rate, which results in lower call quality, but uses roughly 30Kbps. Many vendors support location-basedcodec decisions that allow you to use G.711 when making local network calls andG.729 when making calls known to transit a lower bandwidth link. The overall goal is to transmit the maximum number of voice calls while maintaining acceptable callquality.



Network resiliencyEnd-users expect reliable telephone service, and a successful VoIP solution requires aresilient and redundant network. When possible, the network should contain redundanthardware, redundant links, and adequate power backups. To meet the needs of a VoIPnetwork, core network and VoIP hardware should run off UPS backups. If your VoIPsolution uses power over Ethernet (PoE), the local network switches will also needpower backups. These switches transmit power to the individual telephones, which willneed power and connectivity during an outage.

A strong network design will often translate to a good VoIP design. Medium andlarge VoIP deployments should involve a close examination of the redundancy at thenetwork’s core. Redundancy measures may include dual Layer 3 switches that sharedefault gateway duties through a standards-based or proprietary protocol. The networkmay also have redundant links from the access layer into the distribution and from thedistribution into the core. Not all networks are built to such standards, but you shouldwork to identify any mitigate any points of failure.

Maintaining a healthy VoIP networkYou must continuously monitor and test the VoIP network to ensure sustained success.Although a properly designed voice network will initially respond well, quality canquickly degrade if problems are not detected and resolved. You should regularly monitor bandwidth usage and each hardware device’s CPU and memory use. UsingVoIP-specific management tools you can also monitor the voice network and catchproblems basic management will miss.

VoIP vendors may also use the converged voice and data networks for problemreporting. On some VoIP systems, end users can submit feedback directly from theirtelephones. You can use this information to catch minor issues on which users may notopen traditional help desk tickets.

An effective patch management process is also critical for VoIP network health. Youmust consistently update VoIP hardware and software as you would any other system.VoIP vendors regularly release code revisions and security updates that you must actupon.

The bottom lineYour overall design and implementation process will greatly affect your VoIP installation’ssuccess. Once you identify the VoIP model that best fits your organization, the specificdesign requirements should fall in line. Networking tools such as VLANs and QoS arerequirements and can easily make or break VoIP call quality. Make your network strongenough to handle both day-to-day problems and occasional catastrophes. Finally, moni-tor and maintain your VoIP network to increase its reliability and effectiveness. When you send voice over the data network, quickly detecting and fixing problem is critical forsuccess. Once deployed, VoIP traffic will likely become the most important informationyour network transmits. You should invest time and effort equivalent to this importanceinto planning, implementing and preserving your VoIP network.



Asuccessful voice over Internet Protocol (VoIP) deployment requires carefulconsideration of and adequate planning for the system’s long-term manage-ment requirements. Your deployment and integration plans must identify the

methods and technologies through which you will monitor, support, and maintain relia-bility across the VoIP network. Voice traffic is considerably more sensitive to networkconditions than normal data traffic and your management plans account for this fact.

Along with VoIP management’s technical requirements, your plan should alsoaddress potential personnel issues and functional task changes. Dedicated telecommu-nications personnel normally administer voice systems and conduct voice-specific tasks.As you integrate VoIP with the IP network, these responsibilities may be shifted to orshared with native data-networking staff.

This download outlines methods and best practices for managing your convergedvoice and data network. Failing to adequately plan and effectively implemented properVoIP administration policies and practices will significantly limit the system’s overallfunctionality, reliability and cost savings.

Managing VoIP and the networkAlthough advanced VoIP administration may require specialized tools, standard net-work management practices and procedures can facilitate basic VoIP management.Likewise, existing network management hardware and software can monitor the VoIPnetwork. The groundwork for successful VoIP communications is laid with establishednetwork administration practices.

Successful network resource management is the cornerstone of effective VoIP man-agement. In the most basic sense, VoIP is merely another application vying for networkresources. Yet most organizations and end users view voice communications as a farmore important system; furthermore, users expect highly-reliable telephone systems.Traditional telecommunications networks have provided consistent service for decadesand new technology must meet the same standard. You must efficiently monitor andmanage network resources to ensure VoIP quality of service (QoS).

Successfully implementing VoIP technology requires that you fundamentally rethinktelephone management. You must now monitor bandwidth usage, router CPU utiliza-tion, available memory, interface-based errors, duplex settings and the like. These prob-lems regularly affect data traffic, but VoIP technology makes voice traffic equally sus-ceptible. For example, a router pegged at 100 percent CPU utilization will negativelyimpact all IP traffic, both voice and data.



Improve VoIP management withthese best practicesBy Thomas Nooning, CCNA, CCDA

VoIP Management Best Practices

Use well-established network management practicesUse tools like SNMP,RMON, and Syslog were appropriateRecognize and plan for end-user VoIP QoSexpectationsRegularly review and act upon monitoring informationImplement an effectivechange control processStandardize network hardware and softwareSeparate voice and data traffic via IP packet identifiersMonitor and troubleshootvoice issues with networktraffic analyzersCollect real-time data withVoIP-specific monitoringtoolsBlend traditional telecom-munications and networkadministration skills



Familiar tools such as SNMP (Simple Network Management Protocol), RMON(Remote Monitoring), and Syslog are good means to manage a converged voice anddata infrastructure. These tools help maintain to overall network stability by warning oferror conditions and providing network performance statistics. To be useful however,you must review and act upon the monitoring information in a timely fashion.

Along with effectively monitoring and managing network performance, adding voicetraffic to your IP network reemphasizes the need for adequate change control. Youshould carefully schedule and track any action that could impact the network. Takingthese precautionary measures will reduce the likelihood of a voice outage and helpquickly resolve the problem should an outage or service degradation occur. You shouldcarefully handle even highly redundant networks, with multiple voice paths. A few-sec-ond failover from one distribution switch to the next may not noticeably affect datapackets, but could cause a voice call to experience choppiness or even be disconnected.

Software and hardware standardization also simplify and improve VoIP manage-ment. For example, you can more easily track security vulnerabilities on a single net-work operating system than on multiple systems. In addition, standardization eases net-work documentation creation and renewal.

Maintaining quality of service (QoS)Converged voice and data networks require higher Quality of Service (QoS) standardsthan data-only networks. VoIP technology allows you run voice traffic through yourexisting IP data network, but you must provide a means to logically separate the voicetraffic and give it a higher level of service. If you fail to separate voice and data traffic,problems can occur when larger data packets flood the network and delay the smallervoice packets. Such problems will wreak havoc with VoIP calls.

Most modern network equipment supports some level of traffic distinction, typicallybased on fields within the IP packets. Voice packets are typically generated with specificindicators that determine how the packets should be treated. The network infrastruc-ture must then appropriately use these indicators. For example, a voice packet may besent with its class of service (CoS) field set to 5. The local router is configured to rec-ognize this field and provide it a priority service level with a guaranteed amount ofbandwidth. On the other hand, data traffic may then be sent with a CoS of 0. Thesame router will queue this data traffic, passing the voice traffic first..

There are many ways to match packets and even more ways to treat them oncedetected. Common marking methods include CoS, Differentiated Services code Point(DSCP), or port number matching. This variety of marking methods can create com-plicate the management process. Whenever possible, marking configurations should bestandardized across the network. You must also ensure, often manually, that markingand matching policies are appropriately applied. Proper QoS is often the differencebetween a functional VoIP implantation and one prone to problems.

Monitoring VoIP as a network applicationOnce you build a strong, healthy network to carry your voice traffic, you can thenfocus your attention on monitoring VoIP as a network application. As with any application, VoIP’s intrinsic properties differentiate it from other systems traversing

the network. Voice conversations move in real-time and cannot utilize packet-basederror correction, like that found in Transmission Control Protocol (TCP). While TCPwill retransmit missed packets (often transparently), voice traffic will suffer obviousproblems when such errors occur. Under such conditions there may be garbled audio,extended pauses, and unexpected call termination.

Network traffic analyzers can monitor and troubleshoot voice issues. Some productswill measure call-setup time, jitter, and delay. Other devices will capture voice packetsand play back VoIP conversation. Products like Network Instrument’s Observer andNetwork General’s Sniffer voice modules provide this capability, as well as the ability todecode most voice protocols and codecs.

In-depth VoIP monitoring applications like Prognosis and NetIQ’s offerings will cal-culate a Mean Opinion Score (MOS) for individual calls. MOS values are calculated byaveraging end-user call quality ratings and are therefore subjective, yet you can useMOS values to create quality baselines. Unlike SNMP and other interval-polling-basedtools, these VoIP-specific monitoring tools run in real-time and can identify problemsas they occur.

You can also regularly update your baseline performance metrics with these morerobust VoIP management applications. Some products even have voice packet genera-tors that can simulate VoIP calls. You can use this virtual traffic to stress test your net-work before initial VoIP deployment or rolling out additional VoIP services.

Job role and responsibility shiftsAs telephones blend into the data network, so too do once separate job functions.Traditional telecommunications specialists have a deep understanding of telephony ter-minology and Public Switched Telephone Network (PSTN) circuits. Network adminis-trators are more familiar with the underlying network infrastructure. The VoIP special-ist will require knowledge from both fields as well as VoIP-specific skills.

VoIP technology also simplifies telephone management. Web-based managementutilities and vendor-neutral techniques are increasingly available. Phone moves, addi-tions, and changes become just another help desk task. Day-to-day management isgreatly simplified and only more complex tasks require a voice engineer.

The bottom lineSuccessful VoIP system management begins with your network infrastructure.Implement standard network monitoring and administration practices to ensure a capa-ble, reliable base for the VoIP solution. Use VoIP-specific tools when necessary toincrease QoS and decrease integration problems. Consistent, standardized change man-agement policies and configuration practices will also increase QoS. Finally, blend thespecialized skills of traditional telecommunication personnel and network administra-tors. By nature, VoIP is more dynamic and versatile than legacy telephone systems, butoverall success depends on effective management.



ShoreTel Phone SystemThe ShoreTel phone system scales seamlessly from 0 to 10,000 users making it a perfect match for small, medium and large businesses. A completely integrated system with PBX, voice mail and automated attendant functionality, the solution has installation, administration and management tools that are second to none in the industry that will liberate you from the tyranny – cost and complexity – of other vendor solutions – cobbled together by data networking vendors or repainted to looklike new by legacy-minded vendors. The system is ideal for multi-site companies sincethe distributed architecture unifies multiple locations so their phone system behaves as one, unified system.

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TechRepublic Real World GuideMaximizing Voice over InternetProtocol (VoIP) Networks

Eliminate communication boundariesWith a single phone system spanning multiple locations, enterprises can radically simplify employee communication and boost customer satisfaction. Enterprises can uncover hidden productivity drains as employees use four-digit dialing to reachcoworkers and seamlessly transfer, conference, pick up, park and intercom between sites. Workers can even move locations and assign their extensions to any internal or external telephone with the Office Anywhere™ feature. Customer satisfactionincreases as calling parties connect with the right people faster.

Confidence through reliabilitySystem administrators can rest easy knowing ShoreGear voice switches deliver 99.999% reliability and use an embedded, real-time operating system. A unique callcontrol architecture enables switches to communicate with each other and distributecall processing on the network. If a switch goes offline, its peers automatically compensate, ensuring uninterrupted service. Distributed call processing eliminatessingle-point failures – providing a level of reliability exceeding traditional voice systems.

Smooth migration path, seamless scaleWith five stackable, space-efficient designs, ShoreTel makes it easy to choose the rightsolution for large corporate headquarters, regional branches and small satellite offices.When the organization grows, simply add more ShoreGear voice switches - there areno costly hardware breakpoints. Enterprises can also migrate to IP telephony over timeusing the ShoreGear-T1 and ShoreGear-E1 to provide tandem trunking and coordinat-ed dialing with existing PBXs.

Ergonomic design for effortless communicationShorePhone IP telephones are designed to please the eye as well as the ear. The con-cave sweep of the face places the keys on a horizontal plane while keeping the displayvertically aligned for easy viewing. A bright, backlit display (IP 560 / 530) is easy toread, and the message-waiting light is visible from a full 360° viewing angle. Color-litline buttons (IP 560) provide immediate, at-a-glance information about incoming callsand messages. Designed to conform to the human body, the IP telephone encouragesinteraction. Tactile keys are comfortable to the touch and reduce strain on the fingersand wrists. The precision balanced, contoured handset includes a cushioned grip andfinger notch that makes it comfortable to hold or rest on the shoulder. ShorePhone IPtelephones offer a wideband audio codec that supports seven full octaves of humansound — far superior to the three-octave capability of other vendors’ phones. Highfidelity, full-duplex speakerphones (IP560 / 530 / 210) deliver audio with astonish ingclarity — adding immediacy and depth to hands-free conversation.



Rapid deploymentPlug-and-play installation means enterprises can add new ports and users by simplyconnecting switches to the network. ShoreWare™ Director automatically discoversnew switches and adds them to the ShoreTel system.

Pristine voice qualityIn independent rankings, ShoreTel consistently earns top marks for superior IP teleph-ony call quality. The low latency and toll-quality voice delivered by ShoreGear switchesare the result of ShoreTel technology leadership in dynamic echo cancellation and jitterbuffering, as well as lost packet handling.

Key ProductsThe ideal solution for a headquarters, regional and branch offices, ShoreGear-120/24,ShoreGear-60/12 and ShoreGear-40/8 support up to 120, 60 and 40 IP phones or 24,12 and 8 analog devices respectively. The switches provide an audio input port for amusic-onholdsource, plus an audio output port for overhead paging and night bell services. A power-fail transfer port ensures dial tone during power outages.

ShoreGear-T1 provides a T1 interface for high-density trunking to a central office(CO). ShoreGear-T1 supports loop start, wink start or Primary Rate Interface (PRI)signaling. ShoreGear-T1 can also function as a voice-over-IP gateway to PBX installa-tions – bridging the ShoreTel system to pre-existing legacy systems and smoothing themigration path to IP telephony.

ShoreGear-E1 provides an E1 interface using Euro-ISDN PRI signaling for interna-tional applications. All the ShoreGear switches include two LAN ports for redundantnetwork connections.

Key FeaturesEmbedded call controlShoreGear voice switches are highly reliable devices since they use flash memory forstoring information rather than a mechanical hard disk prone to failure. And since theswitches use VxWorks, the leading embedded, real time operating system from WindRiver Systems, your phone system will not be subject to the numerous attacks andviruses associated with server-based solutions. With embedded call control, you haveconfidence since dial tone is being delivered by the most reliable, robust platform onthe market.

Distributed call controlCall control on the ShoreTel system is distributed to each and every ShoreGear voiceswitch on the network, eliminating any single point of failure. In the unlikely event aShoreGear voice switch fails or becomes isolated by a network fault, the other switcheson the network continue to operate without being affected.



IP phone failoverIf a ShoreGear voice switch supporting IP phones fails or is isolated by a networkfault, the phones will automatically failover to another voice switch providing completeredundancy. And since call control is distributed, redundancy is delivered cost effective-ly with N+1 rather than Nx2configuration required by centralized systems. Second, third, forth level redundancy canbe configured by simply adding additional voice switches.

Gateway failoverIf a ShoreGear voice switch connected to the PSTN fails or is isolated by a networkfault, the system will automatically route calls out an alternative switch.

PSTN failoverIf the wide area network is down, or if admission control for voice traffic on the widearea network is reached, extension to extension calls between sites can automaticallyroute over the PSTN ensuring seamless communication.

Ethernet port failoverShoreGear voice switches feature redundant network uplinks. If the upstream networkdevice fails, the ShoreGear voice switch will automatically failover to the redundant linkensuring continuous operation.

Power failoverThe ShoreGear-120/24, 60/8 and 40/8 all feature power fail transfer. In the event of acomplete power outage that exceeds the duration of your reserve power, one analogtrunk on the voice switch will automatically connect to one analog telephone providingemergency dial tone.



Documents

Maximizing Voice over Internet Protocol (VoIP) Networks