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  1. 1. CCNA Voice QuickReferenceMichael Your S h o r t Cut t o K n o w l e d g e
  2. 2. As a final exam preparation tool, the CCNA Voice Quick Reference provides a concise review of all objectives on the new IIUCexam (640-460). This digital Short Cut provides you with detailed, graphical-based information, highlighting only the keytopics in cram-style format.With this document as your guide, you will review topics on concepts and commands that apply to Cisco UnifiedCommunications for small and medium-sized businesses. This fact-filled Quick Reference allows you to get all-importantinformation at a glance, helping you focus your study on areas of weakness and enhancing your memory retention of essentialexam concepts.About the AuthorMike Valentine has 13 years of experience in the IT field, specializing in network design and installation. He is currently aCisco trainer with Skyline Advanced Technology Services and specializes in Cisco Unified Communications, CCNA, andCCNP classes. His accessible, humorous, and effective teaching style has demystified Cisco for hundreds of students since hebegan teaching in 2002. Mike holds a bachelor of arts degree from the University of British Columbia and currently holds theMCSE: Security, CCNA, CCDA, CCNP, CCVP, IPTX, QoS, CCSI #31461, CIEH, and C T P certifications. He has completedthe CCIE written exam.Mike was on the development team for the Cisco Unified Communications Architecture and Design official Cisco coursewareand is currently developing custom Unified Communications courseware for Skyline. Mike coauthored the popular CCNAExam Cram, second edition, first published in December 2005, as well as the third edition of that volume published inDecember 2007.A b o u t t h e T e c h n i c a l EditorDenise Donohue, CCIE No. 9566, is manager of Solutions Engineering for ePlus Technology in Maryland. She is responsiblefor designing and implementing data and VoIP networks and supporting companies based in the National Capital region. Priorto this role, she was a systems engineer for the data consulting arm of SBC/AT&T. Denise was a Cisco instructor and coursedirector for Global Knowledge and did network consulting for many years. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  3. 3. C C N A Voice Q u i c k Reference by Michael ValentineIntroduction Introduction Voice over IP (VoIP) is no longer an interesting sidebar technology; it is a fact of day-to-day life for millions of people, some of whom are not even aware they are using it. Cisco has aggressively pursued the development and deployment of its Unified Communications suite of products and can now offer an integrated voice, video, and data solution for any business, whether it has just a few employees or a hundred thousand worldwide. The technology is reliable, user friendly, and exciting, but it is not simple—and a successful deployment requires that the designers, implementers, and administra- tors of a Unified Communications system know what they are doing. Training and certification of key staff are strategic components of any business plan to deploy a Unified Communications system. Until recently, the training track for Unified Communications went from the C C N A (the Associate-level routing and switching certification) straight to C C V P , the Professional-level voice certification. The transition between the certifi- cations was difficult for many, because the C C N A did not examine any Unified Communications topics, and the C C V P launched directly into advanced V o I P signaling protocols, Unified Communications Manager administration, traditional telephony, gateway and gatekeeper configuration, Q o S , and so on—all the while assuming that the student had a firm grasp of routing and switching concepts. I have met many good C C N A students who had no telephony or V o I P back- ground and consequently had great difficulty in the C C V P program. Likewise, many students with very strong traditional telephony experience were quickly lost in the intensive data concepts of the C C V P curriculum. It was clear to me and to many of my colleagues that the C C N A was not a good fit as a prerequisite to C C V P . All this brings us to some good decisions that were made regarding Cisco Unified Communications training and certifica- tion. The C C N A has itself been split into C C E N T and C C N A , with C C N A serving as the foundation to some new and specialized certifications at the Associate level. The I I U C curriculum prepares students for the C C N A Voice certification, which in turn is a solid preparation for and a much-needed transition to C C V P . © 2 0 0 8 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  4. 4. CCNA Voice Quick Reference by Michael ValentineIntroduction P u r p o s e of T h i s G u i d e This document serves as a roadmap of the CCNA Voice curriculum and a quick reference for the concepts and commands that apply to Cisco Unified Communications for small and medium-size businesses. This document is not a list of all the questions you may be asked on the exam, but you can be sure that the exam will touch on all the topics you find here. Reviewing this document should help you remember key points and commands you will need to know for the exam. W h o Should Read T h i s G u i d e Anyone who is preparing to take the CCNA Voice exam will find this guide useful. Some may use it as in introduction, and some as a refresher right before their test, some perhaps both. Data networkers who need a quick but complete intro- duction to Cisco Unified Communications for a small or medium-size business will find it useful as well. Those of you who are getting back into study mode for a C C V P exam may turn to this guide as a refresher, too. Then there are always those who simply want to learn something new. Whoever you are, welcome and enjoy the text. I hope you find it useful. Introduction to Unified Communications Todays work environment can be very different from what our parents experienced. The business environment is more competitive, with an unrelenting pressure to be more efficient, to react quickly, and to make important decisions instantly. Efficiencies can be gained by reducing costs, which in turn increases profit, but significant gains can also be made by investing in the business infrastructure so that productivity increases dramatically. Increased productivity means more opportunities to profit from a newfound competitive edge. This is known as Return on Investment, or ROI. The goal is to maximize the ROI—for every dollar spent, businesses want to see more dollars earned, or at least fewer dollars wasted. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  5. 5. CCNA Voice Quick Reference by Michael ValentineIntroduction One area in which businesses have found ways to improve their ROI is in their communications. The evolution of communications from traditional telephony, through cell phones, to smart phones and email, and now to Unified Communications, has created opportunities for businesses to access information and get it to workers instantly. Unified Communications puts voice, data, and video on a converged single network. This makes monitoring, administering, and maintaining the network simpler and more cost effective than if three separate systems existed. Unified Communications also puts powerful applications with information-distribution features right where they are needed. Workers today can be almost anywhere and can carry out meaningful or even critical tasks anywhere they can get a connection to the converged network. The next significant feature of a Unified Communications system is that it is easy to scale, adding more users, more loca- tions, and even more features. Because the Cisco Unified Communications system is a distributed collection of devices, functions, and features that are linked by common protocols, adding a new component is much simpler, and integration of the new components capabilities and features can appear seamless to the people who use the system. The components required to create and use such a system are numerous and complex. Cisco has taken significant steps to develop, document, release, and support the various components as an integrated system. The next section examines the components of a Unified Communications system and introduces the devices and applications that make up the system. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  6. 6. FIGURE 1The UnifiedCommunicationsArchitecture © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  7. 7. • Infrastructure Layer: This layer refers to the network itself, made up of connected switches, routers, and voice gateways. This is the converged network that carries data, voice, and video between users on the system. • Call-Processing Layer: This layer manages the signaling of voice and video calls. When a user picks up the phone and dials a number, the call processing agent determines how to route the call, instructs the phones to play dial tone or to ring, and records the details of the call for future analysis. The call agent carries out many other functions; it can be considered the equivalent of a traditional PBX system, but with many more features. • Applications Layer: This layer features elements such as voice mail, call-center applications, billing systems, time- card or training systems, and customer resource management applications—to name just some of the many applica- tions that can integrate with, draw from, or otherwise complement the Unified Communications systems. Because the Unified Communications systems are distributed (meaning not constrained to one box or even one location), the applications can be hosted almost anywhere, given appropriate connectivity. • Endpoint Layer: This layer includes the parts of the system that the users see, hear, or touch. This includes Cisco Unified IP Phones, PCs with software phones, video terminals, or other applications that send and receive informa- tion from the Unified Communications system.The following sections examine the layers in a little more detail.Infrastructure LayerAt the infrastructure layer, we are building the connections between all the devices that send and receive data, voice, andvideo. These include Layer 2 and 3 switches, routers, and voice gateways. Voice gateways are among the most importantcomponents because they provide the connection to the PSTN or other network carriers. One of the critical functions (andone that is unfortunately often underemphasized in many deployments) is quality of service, or QoS. QoS providesservice guarantees to various types of network traffic, in particular voice and video traffic. Without QoS, you can experi-ence poor call quality or even failed calls. Infrastructure design and deployment is literally the foundation of the system; © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  8. 8. if any weaknesses exist here, they will manifest as system failures or unreliability. It is very important to build a solid andcorrect foundation. The goal is to achieve 99.999% uptime; achieving that goal takes careful attention and good design.Call Processing LayerThe call processing layer is chiefly about the call agents. A call agent is not a person; it is an application that looks at thesignaling traffic from devices that place and receive calls, and it determines what to do with the call. A Unified IP Phonesends a packet to the call agent when you lift the receiver; the call agent instructs the phone to play a dial tone. When youbegin dialing a number to call, the call agent receives the digits and tries to find a match for the number in its dial plan. Ifthe destination number is a phone that it controls, it tells the called device to ring. During the call, the call agent also setsup other services, such as Hold, Call Park, Transfer, Conference, and so on. The call agent also instructs the phones totear down the call when one party hangs up. The call agent usually keeps detailed records of each call made; these arecommonly used for billing purposes or troubleshooting.Cisco provides several options for call agents, matched to the size and requirements of the customer: • The Cisco Smart Business Communications System is designed for small businesses with up to 48 users. The system runs on the Cisco Unified Communications 500 Series for Small Business devices. • Cisco Unified Communications Manager Express serves up to 240 users and runs on the Integrated Services Router platforms. • Cisco Unified Communications Manager Business Edition handles up to 500 users and runs as a standalone installa- tion on a 7800-series Media Convergence server. • Cisco Unified Communications Manager can handle 30,000 or more users and runs on clusters of 7800-series Media Convergence servers. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  9. 9. Smart Business Communications SystemFIGURE 2The Smart BusinessCommunicationsSystem—Image ©Cisco The Smart Business Communications System is a group of specially designed, integrated devices that can provide high- quality routing, firewall, intrusion prevention, Power over Ethernet, wireless, and many WAN and PSTN connectivity options. It is essentially a solution-in-a-box, with a simple web-based interface that is largely plug and play. The Unified Communications 500 Series devices are small and inexpensive, providing the kind of connectivity options small busi- nesses need to allow them to take advantage of Unified Communications with a good ROI. The SBCS is expandable using 500-series switches, and the call agent software can support up to 48 phones. Voice mail and Auto-Attendant func- tions are provided by the integrated Cisco Unity Express application. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  10. 10. Unified Communication Manager ExpressFIGURE 3Cisco IntegratedServices Routers forUnifiedCommunicationsManager Express—Image © Cisco Cisco Unified Communication Manager Express is a software feature that can run on the ISR-series router platforms, including the 800, 1800, 2800, 3800, and 7200-series platforms. The call agent application is embedded with the Cisco IOS software and is configured either from the command line or a Web-based interface. Unified CM Express is a full- featured call agent that is cost-effective, reliable, and scalable and integrates with both Service Provider connections and Unified Communications Manager clusters. With support for both H.323 and SIP protocols, site-to-site connections are possible in a variety of environments. The Unified CM Express system can be set up either as a PBX or a Key switch system, providing customers with a familiar experience that suits their operating environment. Unified Communications Manager, Business Edition Unified Communications Manager, Business Edition is a standalone installation of the Unified CM application and Cisco Unity Connection, coresident in a single M C S 7800-series appliance. This system can support up to 500 users in a single site or multisite centralized deployment and can be migrated to a full CM cluster if growth necessitates it. Unified CM Business Edition provides medium-size businesses with advanced features such as Mobility (a.k.a. Single Number © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  11. 11. Reach), Do Not Disturb, Intercom Whisper, and Audible Message Waiting Indication, as well as speech recognition andintegrated messaging. Because Unified CM Business Edition uses the same call agent software as a full cluster deploy-ment of Unified C M , it supports full integration with the other Unified Communications applications, such as UnifiedPresence, Unified Personal Communicator, MeetingPlace Express, Contact Center Express, and so on.Unified Communications ManagerThe full version of Unified Communications Manager is an enterprise-class, fully scalable, redundant, and robust distrib-uted packet-telephony application. Scalable to 30,000 users per cluster, with the capability to form intercluster connec-tions, it can support a global unified communications system for hundreds of thousands of endpoints. Unified CMversions prior to 5.x are Windows based, whereas versions 5.x and 6.x are Linux-based appliances.Applications LayerThere are effectively a limitless number of applications that can be part of a Unified Communications system, becausethird-party applications can be developed to closely integrate with the Cisco suite of products. The following is a list ofthe more common applications found in a Unified Communications system: • Voice Mail: Voice mail can be provided using Cisco Unity, Unity Connection, or Unity Express. Unity and Unity Connection run on the MCS 7800 series platforms, and Unity Express is a self-contained module that is added to an ISR router and administered through the command line and GUI. The maximum mailboxes and recording time capacities vary depending on which module (either Advanced Integration Module or Network Module) is installed in the router. • Cisco Emergency Responder: This application tracks the location of an IP telephony device based on the physical switch port it is connected to. This information is attached to the caller information in the event the device calls 9 1 1 , which in turn allows 911 responders to locate the device (and therefore presumably the emergency) more precisely. 911 operation in a Unified Communications environment is a major design challenge because a VoIP phone system can easily throw out the premise that a PSTN call is placed from the same location as the phone that made it. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  12. 12. • Cisco Unified Contact Center [Express]: This is a call center application with full feature support for advanced call distribution, supervision, escalation and logging. Versions are available to support small and large call centers. • Cisco Unified Meeting Place [Express]: This is a full-featured web-conferencing application enabling voice and video conferencing as well as document sharing and collaboration, whiteboarding, and conference participant management. • Cisco Unified Presence: This extends the native capabilities of Unified CM 6.x+ to indicate presence information. The native capability includes on/off hook status in speed dials and call lists, whereas the full applications server provides detailed presence information as typically found in chat applications ("On the Phone," "Out to Lunch," "Do Not Disturb," and so on).Endpoints LayerAn increasing variety of Cisco Unified IP Phones (and third-party IP phones) can be part of a Unified Communicationsdeployment. All Cisco Unified IP Phones provide a display-based user interface, user customization, Power over Ethernetcapability (where appropriate), and support for G.711 and G.729 codecs (and, on some models, Cisco Wideband and/oriLBC codecs). The following is a partial list and brief description of the Cisco Unified IP Phones available:Commercial/Retail Phones 7931G: 24 programmable buttons, 4-way LEDs, Dedicated HolaVTransfer/Redial buttons 7921G: Wireless, 2-in. color screen, speakerphone, XML-PTT, longer battery lifeMobility 7921G: Wireless, 2-in. color screen, speakerphone, XML-PTT, longer battery life IP Communicator: Software-based IP Phone, emulates 7970G functionality © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  13. 13. Business Class 7940G: B/W LCD, 2-button, XML-capable, SIP-capable 7941G: Higher resolution B/W LCD, 2-button, XML-capable, SIP-capable 7960G: B/W LCD, 6-button, XML-capable, SIP-capable 7961G: Higher resolution B/W LCD, 6-button, XML-capable, SIP-capableAdvanced Media 7942G: Hi-fidelity audio, Hi-res display, 2-button, XML-capable, SIP-capable 7945G: Gig Ethernet, Hi-fidelity audio, Hi-res color display, 2-button, XML-capable, SIP-capable 7962G: Hi-fidelity audio, Hi-res display, 6-button, XML-capable, SIP-capable 7965G: Gig Ethernet, Hi-fidelity audio, Hi-res color display, 6-button, XML-capable, SIP-capable 7975G: Gig Ethernet, Hi-fidelity audio, backlit hi-res color display, 6-button, XML, SIP-capableColor Touch 7970G: Backlit hi-res color touch screen, 8-button, XML-capable, SIP-capable 7971G-GE: Gig Ethernet, Backlit hi-res color touch screen, 8-button, XML-capable, SIP-capable 7975G: Hi-fidelity audio, Backlit hi-res color touch screen, 8-button, XML-capable, SIP-capableVideo 7985G: Personal desktop video phone Unified Video Advantage: Software IP Video Phone with support for attached cameraConference 7936G: Backlit LCD, 3 softkeys, small-medium conference needs 7937G: Hi-fidelity audio, extended audio coverage w/ extra mics, large display © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  14. 14. Understanding Unified CommunicationsApplicationsIn this section, we examine the variety of applications available for integration in a Unified Communications environ-ment, including Messaging, Auto Attendant, Interactive Voice Response (IVR), Contact Center, Mobility, and Presence.MessagingA variety of messaging options are available to suit the needs of businesses small and large. The following table providesa summary of the options. Max. T D M PBXProduct Users Messaging Capability Platform Integration? Networking? Redundancy?Unity Express 250 Voice Mail + Integrated Messaging ISR No Yes NoUnity Connection 3000 Voice Mail + Integrated Messaging MCS Yes No NoUnity 7500 Voice Mail + Integrated Messaging + MCS Yes Yes Yes per server Unified MessagingThe following sections describe the messaging products listed in the table in more detail.Cisco Unity ExpressUnity Express is an ISR-based application that runs either on an AIM module or an NM module. AIM modules areconnected to the main board as a daughter board addition and use flash memory for greetings and message storage. AIMmodules therefore have less capacity for storage. NM modules are inserted into module bays in ISR routers, use a hard © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  15. 15. disk for greeting and message storage, and have greater capacity for storage than A I M modules. Unity Express supportsfrom 4 to 16 concurrent sessions and 12 to 250 mailboxes (dependent on the module and platform installed). UnityExpress is managed through the command line or a web-based GUI. It allows users to view and sort their voice messagesusing the IP Phone display, email application, or IMAP client. Unity Express can be deployed in conjunction withUnified CM or CM Express and can supplement a full Unity deployment.Cisco Unity ConnectionUnity Connection is a medium-size business solution with a full range of messaging features. It can be deployed on itsown or as a coresident installation as part of Unified Communications Manager Business Edition on suitable MCS plat-forms. When deployed as part of CM Business Edition, Unity Connection supports up to 500 users; when deployed as astandalone application, Unity connection supports up to 3000 users per server (dependent on hardware). Scalability isachieved by networking up to 10 other Unity messaging products of any type. Fourteen languages are supported fordeployments worldwide. Unity Connection also supports speech recognition, allowing users to speak commands tomanage their messages hands-free. Multiple interfaces are supported for managing messages from an IP Phone, an emailclient, a web GUI, or Cisco Unified Personal Communicator. Users can define their own rules to transfer calls based oncaller, time of day, and Microsoft Exchange calendar status.Cisco UnityUnity is the enterprise-class messaging application with support for up to 7500 users per server and up to 250,000 usersin a multi server networked environment. Interoperability with legacy voice-mail systems, notably Octel and Nortelsystems, allows a phased transition to IP messaging with minimal disruption to users. Unity supports 35 languages, facili-tating deployments worldwide. Full unified messaging is possible with connectors for Exchange, Notes, and GroupWise,providing a single inbox for email, voice mail, and fax messages. Text-to-speech capability allows users to have theiremails read to them over the phone by the RealSpeech engine; speech recognition is also available so users can instructUnity to play, search, or record messages hands-free. Secure messaging is supported, allowing encrypted messages andpreventing messages that have expired from being played. Access to messages is made simple, intuitive, and possiblefrom almost anywhere. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  16. 16. Auto AttendantAn Auto Attendant is basically an advanced answering machine; instead of only one message, it can play several, depend-ing on the date and time, which number was called, and most importantly, what numbers the callers pressed in responseto the greeting they heard. If you have ever heard: "For service in English, press I. Pour service en Francais, appuyez surle 2 . . . , " you have been served by an Auto Attendant. Typically, Auto Attendants allow callers to select the department orextension they want to call, and often they allow the caller to spell out a first or last name to search in the company direc-tory. Cisco Unity, Unity Connection, and Unity Express all provide Auto Attendant functionality; Unity and UnityConnection include a simple web interface that makes it very easy to construct menus and test to see that they work asyou intended.Cisco Unified IP IVRAlthough Auto Attendants are useful, their functionality is limited to pretty basic menu navigation. To scale this function-ality up to call-center size, and especially to include speech recognition, prompt-and-collect ("Please enter your 10-digitaccount number, followed by the # sign"), Text-to-Speech, database integration, and Java application integration, a muchmore advanced IVR application is required. Cisco Unified IP IVR has all these advanced capabilities. Call centers thathave a high call volume and many possible queues of callers waiting for different agent capabilities can effectively deployUnified IP IVR to steer callers to the correct agent, or perhaps to an automated information source without the need to tieup an agent at all. Unified IP IVR includes the capability to provide both real-time and historical reports on its utilizationand offers multiple-language support.Cisco Unified Customer Voice PortalFor the very largest call centers, the Unified CVP product provides advanced IVR, including speech recognition,advanced queuing, integration with Cisco Unified Contact Center (Enterprise and Hosted), and powerful call routing,management, and reporting features. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  17. 17. Cisco Unified Contact CenterCisco provides a range of Contact Center products for S M B , Enterprise, and Service Provider applications. Customercontact solutions provide multiple avenues to reach and interact with customers, including basic telephony as well asfeature-rich web, email, and even video interaction. The three Contact Center products are described next: • Cisco Unified Contact Center Express: Suitable for 10 to 300 agents, it provides sophisticated call routing, outbound dialing capabilities, comprehensive contact management, and chat and web collaboration in a single- server, integrated "contact center in a box." • Cisco Unified Contact Center Enterprise: Provides intelligent contact routing, call treatment, network-to-desktop computer telephony integration (CTI), and multichannel contact management. It combines multichannel automatic call distributor (ACD) functionality. Sophisticated monitoring allows customers to be routed to the most appropriate agent (based on real-time conditions such as agent skills, availability, and queue lengths) anywhere in the enterprise, regardless of the agents location. • Cisco Unified Contact Center Hosted: An application hosted by service providers, who then lease its functionality to customers who want a virtual contact center without the need to manage and maintain it themselves. Subscribing business customers can have IP or time-division multiplexing (TDM) infrastructures or a combination of the two. Contact Center Hosted provides all the advanced capabilities found in Contact Center Enterprise.Cisco Unified Mobile SolutionsTodays workforce is mobile, distributed, and utilizes multiple technologies to communicate. The desire to have a seam-less transition between the various ways in which people can be reached has spurred the development of mobility featuresin Cisco Unified Communications. The key products are the following: • Cisco Unified Mobility: (a.k.a Single Number Reach) Allows multiple remote destinations (commonly a cell phone, a home office phone, or other work location) to be configured to ring at the same time as the workers enterprise © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  18. 18. desk phone. Thus, when a customer calls your work number while you are on your way to a meeting, your cell phone can ring and you can answer without the customer realizing you are away from your desk. Furthermore, if you return to your desk, you can simply pick up your desk phone and continue the call. A related feature, called Cisco Mobile Voice Access, allows users to place calls from their enterprise desk phone from a remote location or a cell phone. By dialing a configured number and entering an access code, the enterprise system will prompt for the number you want to call, and the call will be placed as if you were at your desk. This is useful not only for presenting the preferred Caller-ID number to the customer, but also potentially for long-distance toll savings.• Cisco Unified Personal Communicator: A desktop PC (or Mac) application that combines a software IP Phone, IM client, video, and online collaboration capabilities. Presence indications ("Busy," "In a call," "Away," "Do Not Disturb," and so on) can save time and enhance productivity because users can see the status of the person they want to contact before trying to reach them. Integration with an Outlook toolbar provides click-to-call or click-to-chat from a message or contact.• Cisco Unified IP Communicator: A fully functioned software IP Phone, often characterized as a "7970 under glass." Users can place and receive calls from their PCs from anywhere that connectivity to the call agent can be established. This is typically achieved through a VPN connection; it is perfectly possible to place a call from an airport boarding lounge or your local coffee shop. Unified IP Communicator can be enhanced with Unified Video Advantage, which integrates a PC webcam for video calls.• Cisco Unified Mobile Communicator: An application for smart mobile phones that provides access to enterprise directories, presence indicators, secure text/chat, voice-mail access, call history of any of the users phones displayed on the mobile handset, and collaboration and conferencing integration with Unified Meeting Place.• Cisco Unified Presence: A server-based application that extends the on/off hook status monitoring capability of Unified CM 6.x to include IM-like status messages. Status indications can be displayed or integrated with Personal Communi- cator, Mobile Communicator, IP Phone Messenger, the Microsoft Office Connector, and IBM Sametime Communicator. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  19. 19. nified Communications Applications Cisco Telepresence Cisco Telepresence is a state-of-the-art high-definition videoconferencing system. A specially designed system of furniture, cameras, monitors, and microphones creates a life-sized illusion of a meeting whose participants may be half a world apart. With 1080p HD video, CD-quality spatial audio, and high-quality lighting, the experience is dramatic to say the least. In combination with the Telepresence Multipoint Switch, up to 36 locations can be included in a single conference with near- zero latency. This can only be described as a high-end solution, with commensurate demands on bandwidth.FIGURE 4The CiscoTelepresence 3000System—Image ©Cisco © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  20. 20. Understanding Traditional Telephony This section introduces traditional telephony systems, concepts, and applications. The PSTNFIGURE 5 Public Switched Telephone NetworkA Representation ofthe Public SwitchedTelephone Network(PSTN) The PSTN, or Public Switched Telephone Network, is made up of Central Office switches to which subscriber lines are connected. The CO switch is programmed so that it knows which phone number (subscriber line) is attached to a particu- lar port. If the number called is not on the local switch, the call is routed over an interoffice trunk to another switch, which may have the called subscriber line connected directly to it or may in turn route the call to other CO switches. Telephone numbering plans are organized so that calls are routed efficiently through the switch system to the correct destination switch. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  21. 21. Note that for our purposes, a line connects to a single phone number and supports one call at a time, whereas a trunk interconnects two switches and supports multiple calls at a time. Business Telephony Systems Businesses have more elaborate requirements of the telephone beyond simply placing calls. Over time, two main types of business systems have evolved: the PBX and the Key System. Both have their place, and both offer calling features that make it easier to carry on business both internally and externally with staff, customers, and suppliers. PBX SystemsFIGURE 6A Representation of aPBX System Business telephone systems often use a Private Branch Exchange (PBX) switch, usually located in their building. The PBX is configured in much the same way as the PSTN CO switch: it holds the dial plan for all numbers within the busi- ness, and external calls are routed over a CO trunk to the PSTN CO switch if the called number is not on the PBX. As a business grows, it is common to install another PBX in another location or building and set up a special trunk (called a tie-line or tie-trunk) between the PBXs so that calls to the remote location are still internal numbers (typically 4- or 5- digit numbers) instead of PSTN calls. A PBX consists of a control plane (the "brain"), a terminal interface that connects phones to the features they want to use, a switching engine that determines which port to route a call out, line cards to connect to phones, and trunk cards to © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  22. 22. CCNA Voice Quick Reference by Michael ValentineUnderstanding Traditional Telephony connect to the PSTN or to tie trunks to other PBXs. PBXs come in a variety of sizes, supporting from 10 to 20,000 phones. All PBXs offer basic calling features, with additional advanced features optional based on hardware capability and licensing. These features typically include Hold, Transfer, Conference, Park, Voice Mail, and so forth. Key Systems Smaller businesses will sometimes use a key system. A key system is like a PBX in that it controls a group of local phones, but key systems tend to have fewer features than PBXs. One characteristic of key systems that many businesses specifically request is distributed answering from any phone; that is, all the phones ring at once, and whoever is able to pick up Line 2 (for example) can push the Line 2 button on any phone and take the call. PBXs dont normally do this; they have a central answering point (a receptionist or Auto Attendant) and Direct Inward Dial numbers (DIDs) if needed. Telephony Signaling Telephony signaling refers to the messages that must be sent to set up and tear down a phone call—that is, anything other than the actual voice. Following are the three types of telephony signaling: • Supervisory: Communicates the current state of the telephony device. There are three types of supervisory signals: • On-Hook: The phone is hung up. Only the ringer is active in this state. (Note that if the speakerphone button is pressed, this is the same as being off-hook.) • Off-Hook: The phone receiver is out of the cradle. This signals the phone switch (PSTN, PBX, or Key) that the phone wants to make a call; the switch sends a dial tone to indicate that it is ready to receive digits. • Ringing: The switch sends voltage to the phone to make it ring, alerting the user that there is an inbound call. The other end of the call hears a ringback tone. • Address: Communicates the digits that were dialed. Address signaling is most commonly done using Dual Tone Multi Frequency (DTMF) tones, commonly known as TouchTone dialing. The combination of tones tells the switch what number was pressed. Older systems also support pulse dialing, which is what the old-fashioned rotary dial © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  23. 23. CCNA Voice Quick Reference by Michael Valentine Understanding Traditional Telephony phones used. Pulse dialing works by repeatedly opening and closing the circuit to the phone switch; the switch counts the number of pulses and interprets that as the number dialed. You might have seen in really old movies when someone picks up the phone and taps the receiver cradle repeatedly; this was how you got the attention of the operator. • Informational: Communicates the call status to participants in the call. Informational signals include dial tone, ring- back tone, and reorder tone. These tones, and others not mentioned here, will vary from country to country. In England, for example, ringback tone sounds very different from what would be heard in North America. Signaling System 7 (SS7) SS7 is a global telephony standard that allows a phone call to be routed between CO switches, between long-distance carriers, and even between national telephone providers in other countries. SS7s primary role is to complete the setup and teardown of phone calls; this is quite a distinct process from the actual transport of the voice signal. In fact, the call control information in an SS7 network must traverse an entirely separate network from the voice path. The capabilities of SS7 have allowed the introduction of relatively complex value-added services, such as call screening, number portability, and prepaid calling cards. PSTN Call Setup To make a PSTN call, several steps occur that the caller is unaware of. The following steps refer to Figure 7. 0FIGURE 7PSTN Call Setup Customer Telephone © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  24. 24. 1. The calling phone goes off-hook, closing the circuit to the local CO switch. 2. The local CO switch detects that current is flowing over the closed circuit and sends a dial tone to the calling phone. 3. Address signals (DTMF or pulse) are sent as the calling party dials the called number. 4. The local CO switch collects the digits and makes its routing decision; in this example, it uses an SS7 lookup to locate the destination CO switch. 5. Supervisory signaling indicates to the far-end trunk that a call is inbound. 6. The PBX determines which internal line the call should go to and causes the connected phone to ring. 7. The ringback tone is heard at the calling party end. 8. The called party goes off-hook, and a voice circuit is established end-to-end. The fact that all this happens with very high reliability billions of times every day is pretty impressive. It also provides some insight into how complex it is to duplicate these functions in a VoIP system. More on that later. Numbering PlansNOTE A numbering plan is an organized distribution of telephone numbers administered by a regional or national authority. TheCodes do not always plan defines the rules that allocate numbers according to an established international telecommunications standard. Forneed to be dialed; Local example, the North American Numbering Plan defines a country code of 1, followed by a three-digit area code, a three-numbers must always be digit office code, and a four-digit local number. There are numerous other numbering plans for other countries or regionsdialed. of the world. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  25. 25. The North American Numbering Plan Lets look at the N A N P in more detail. The 10-digit number is made up of the 3-digit area code, the 3-digit office code, and the 4-digit local number, as shown here: NXX-NXX-XXXXNOTE It is very important to note that the " N " represents any digit in the range 2 through 9, and the " X " represents any digit 0Several other ranges are through 9. You will never find an office or area code of OXX or 1XX; those numbers are either reserved for specializedreserved for specialized purposes or would interfere with things like operator access numbers. Several ranges are also reserved for Easilypurposes. One commonly Recognizable Codes (ERCs); these are numbers where the second and third numbers of the area code are the same. Theyrecognized one is 555- are used for special services—for example, 800, 888, 877, and 866 are toll-free numbers. Another recognizable assign-01XX, which is used infilm and TV, demonstra- ment is the " N i l " series: this includes 4 1 1 , 6 1 1 , and 911 numbers that are not used as area codes but for other specialtions, or education. No assignments, such as information or emergency services.actual customer isassigned these numbers,so calling a number seenin a movie will not pose a E.164 Addressingnuisance to anyone. When The E.164 addressing scheme is an international standard for telephone numbering plans, originally developed by theTommy Tutone recorded International Telecommunication Union. An E.164 number contains the following components:"867-5309/Jenny," heimmediately annoyed CC—Country Codethousands of phone NDC—National Designation Codecustomers worldwide. SN—Subscriber Number An E.164 number is standardized at 15 digits, generating over 100 trillion unique strings. In theory, its possible to direct dial any conventional phone in the world from any other conventional phone. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  26. 26. Introduction to Analog CircuitsAnalog (in contrast to digital) circuits are still the most common telephone connections worldwide. The phone line to aNorth American home is most commonly an analog loop circuit, although more and more digital phone services are beinginstalled. Cisco gateways must connect to various analog services to place calls to the PSTN; the analog circuits thatCisco supports are Foreign Exchange Station (FXS), Foreign Exchange Office (FXO), and Earth and Magneto (E&M).This section examines the components of an analog telephone and the signaling methods used by analog circuits.Components of an Analog PhoneAn analog phone includes the following components: • Receiver: The handset speaker • Transmitter: The handset microphone • 2-wire/4-wire hybrid: Converts 2-wire from the CO to 4-wire in the phone • Dialer (tone/pulse): The dialing keypad or rotary dial • Switch hook: The switch that closes/opens the circuit (off-hook/on-hook) • Ringer: Sounds to indicate inbound callForeign Exchange StationAn FXS port connects directly to an analog phone or fax machine. Switches (including CO switches and PBXs) andCisco gateways will have FXS ports to connect an analog phone. The switch or FXS gateway port must provide power,call progress tones, and dial tone to the analog device. An FXS port on a gateway is also the direct connection to the VoIPnetwork and consequently also contains a coder-decoder (Codec) to convert the analog signal to digital for packetization.Alternatively, a Cisco Analog Telephony Adapter can serve as a remote FXS-to-Ethernet converter to connect an analogstation to the VoIP network. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  27. 27. Foreign Exchange Office An F X O port connects to the PSTN CO switch. If you want to connect your gateway router to the phone company over standard analog lines (that you could plug your analog phone into), you use F X O ports. These ports allow the gateway to place and receive calls to/from the PSTN. FXO ports also include a codec.FIGURE 8Loop-Start Signaling Loop-start signaling is commonly associated with local loop circuits (such as an analog line to the PSTN); it is seldom seen on trunk connections. A local loop is a two-wire service that uses very simple electrical signaling; remember that this technology has been in use and substantially unchanged for 100 years! © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  28. 28. Following is the loop-start process: 1. A phone that is on-hook breaks the electrical circuit; we say opens the circuit. No electricity can flow because of the open circuit. 2. When the receiver is lifted, the circuit closes and electricity flows. This current is - 4 8 V DC. The CO switch that is connected to the local loop detects the current flow and interprets this as an attempt to place a call—we say "seize a circuit." The CO switch plays dial tone down the line to the phone as an indication that it is prepared to collect digits. 3. If the phone is on-hook and the CO switch has a call inbound for it, the CO switch applies 90V AC current to the open circuit; because it is AC, the current can be applied even on the open circuit. By the way, this is why you should not have an analog phone near the bath. The DC voltage wont do much, but you will definitely know it if the phone rings and you get zapped by the AC voltage. Loop-start works very well for homes or other lightly used circuits, but if it is in constant use, a problem known as glare can occur; this refers to both ends of the circuit being seized at the same time, so that you pick up the phone and there is a caller on the line at the same moment, by coincidence.Ground-Start SignalingGround-start signaling is an adaptation of loop-start. Instead of the circuit being closed only at the phone end, both endsof the circuit have the capability to detect current, and both ends can request and confirm the use of the circuit. This isachieved by both ends being able to ground one of the wires in the circuit. These wires (or leads) are referred to as Tipand Ring. These terms date back to the use of 1/4-inch jacks with a positive contact at the tip and a negative conductor inthe ring. The advantage is that it makes glare much less likely, and consequently ground-start is appropriate for trunkconnections that are heavily used. However, it is very rare to see a ground-start trunk in a VoIP network or indeed in anynew trunk deployment. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  29. 29. FIGURE 9Ground-StartSignaling The ground-start process as it occurs on a trunk between a PBX and the CO switch is described next; refer to the diagram for each step: 1. The PBX has a call that it must send to the PSTN. It signals to the CO switch that there is an inbound call by grounding the ring lead. 2. The CO senses the ring lead as grounded and grounds the tip lead to signal the PBX that it is ready to receive the call. 3. The PBX senses the tip ground and closes the loop between tip and ring in response; the PBX also removes the ring ground. 4. The voice circuit is complete, and communication can begin. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  30. 30. E&M SignalingVariously called "Ear and Mouth," "RecEive and TransMit," and "Earth and Magneto," E & M analog trunks were typi-cally used to interconnect PBXs (tie-trunks). E & M connections have separate leads for signaling and voice; the signalingleads are known as the E and M leads.In an E & M connection, one side is called the trunk side; this is usually the PBX side. The other side is called thesignaling-unit side; this is the CO, channel-bank, or Cisco gateway E & M interface. The E lead is used to indicate to thetrunk side that the signaling-unit side has gone off-hook; conversely, the M lead is used to indicate to the signaling-unitside that the trunk side has gone off-hook.Five types of E & M signaling exist, numbered Type I through Type V. In a Cisco Gateway application, Types II and V canbe connected back-to-back and Type I cannot be. Cisco does not support Type IV.Three main techniques are employed in E & M circuit signaling: • Wink Start: The terminating side (for example, a Cisco Gateway) uses a brief off-hook-on-hook "wink" to acknowledge that the originating side (for example, a PBX) has gone off-hook. Upon receipt of the wink, the origi- nating side begins sending digits. When the far-end device answers the call, the terminating side goes off-hook and the voice circuit is then set up. • Immediate Start: The originating side goes off-hook, waits a set time (perhaps 200ms), and then begins sending digits whether or not the terminating side is ready. • Delay Dial: Assume that a PBX is placing a call outbound to the PSTN: First, the PBX goes off-hook. The CO then goes off-hook until it is ready to receive digits; it then goes on-hook. (This time period is the delay dial signal.) The PBX sends digits. When the far-end device answers the call, the CO goes off-hook (called Answer Supervision), and the voice circuit is then set up. The advantage of Delay Dial is that some equipment is not ready to receive digits instantly, even though it has sent the wink; the delay compensates for this. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  31. 31. Introduction to Digital CircuitsDigital circuits have the chief advantage of allowing a much higher density of calls on a given physical connection; ananalog circuit can handle only one call at a time, whereas a digital circuit can handle many.There are two main types of digital circuits: Common Channel Signaling (CCS) and Channel Associated Signaling(CAS). CAS circuits are available in two speeds: Tl at 1.544Mbs supports 24 calls, and El at 2.048Mbs supports 30calls. (For these values, we are assuming the calls are not compressed; more on this later). CCS circuits are designated asPRI T l , PRI E l , and BRI. A PRI Tl can support 23 calls, a PRI El 30, and a BRI only 2.The use of a digital circuit by definition implies that the voice signal must be digitized; the conversion from analog todigital is performed by a codec. The following sections discuss the conversion of analog to digital.Digitizing Analog SignalsThere are four steps in the process of digitizing analog sound: 1. Sample the analog sound at regular intervals 2. Quantize the sample 3. Encode the value into a binary expression 4. Optionally compress the sampleSampling could be done any number of times per second; the more samples taken per second, the higher the audioquality, but the amount of digital data produced is much larger. Nyquists theorem states that the sampling interval shouldbe 2x the highest frequency of the sample to produce acceptable audio quality during playback. Because the highestfrequency in human speech that we want to reproduce in telephony is around 4000 Hz, the sampling rate for standard toll-quality digital voice is 8000 intervals per second. By contrast, CD music audio, which must encode both much higher andmuch lower frequencies, samples at about 192,000 times per second. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  32. 32. Quantizing refers to making a digital approximation of an analog waveform. Imagine drawing an arc on a chessboard; ifyou had to define the arc using only the square it was in for each row (segment) and column (interval), you would end upwith a stepped pattern that was sort of close to the original arc but not exact. This is exactly the process that happens withquantization: the codec chooses a segment value that is as close as possible to the analog value at the interval it wassampled, but it cannot be exact. To make the quantization more accurate, each sample is divided into 16 intervals that areadjusted to more closely match the sampled wave. Furthermore, the segments are actually more fine-grained at the originthan at the high and low ranges. This is because most of the human speech we are trying to capture accurately is in thiscenter range of the scale; there are fewer sounds at the very highest and lowest values. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  33. 33. FIGURE 11Quantizing the DigitalSample Encoding the signal is a simple process. We have a single 8-bit code word to identify whether the analog signal was a positive or negative voltage, what value the signal was quantized to (which segment), and finally, which interval is repre- sented by the code word. The first bit identifies either positive voltage (1) or negative (0). The next three bits represent the segment. There are eight segments in the positive range and eight segments on the negative range, so three bits provide the necessary encoding for the quantization. The last four bits identify the interval. A code word example is shown next: 10011100 In this case, the first 1 indicates a positive voltage; the next digits of 001 indicate this is the first segment (on the positive side), and 1100 indicates the twelfth interval. The code word is 8 bits; we generate a code word 8000 times per second (the sample rate). This gives us a bitrate output of 8 x 8000 = 64,000 bps (64 kbps). The process we just described is known as Pulse Code Modulation (PCM) and is the standard for uncompressed digital voice in telephony. One voice stream thus requires 64k of bandwidth for transport. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  34. 34. NOTE Compression is not a required step, but it is often done to save bandwidth in VoIP environments. The two main types ofThe determination of compression we are concerned with are the following:voice quality is based onthe Mean Opinion Score • Adaptive Differential P C M (ADPCM): This method does not send entire code words, but instead sends a smaller(MOS). This is a subjec- code that represents the difference between this word and the last one sent. This is not commonly used today,tive measurement, because it produces lower voice quality and compresses down only to about 16 kbps.created by gathering theopinion of live human • Conjugate Structure Algebraic Code Excited Linear Prediction (CS_ACELP): As the name suggests, this islisteners. A sample more complex compression. Based on a dictionary or codebook of known sounds made by a standardized Americanrecording is played, and male voice, the digital sample is analyzed and compared to the dictionary. The dictionary code that is the closest tothe listeners give it ascore out of 5, where 5 is the sample is sent. The codebook is constantly learning. The output of this compression is typically 8 kbps—withbest. The same sample is very little degradation of voice quality. This compression is widely used in VoIP.played using differentcompression or process-ing methods and scoredagain. Because MOS is Time Division Multiplexing (TDM)so subjective, other T D M is the primary technology used in traditional digital voice; it is also extensively used in data circuits. The basicquality measurementsexist that are more premise is to take pieces of multiple streams of digital data and interleave them on a single transmission medium.empirical and more accu-rate. For reference, stan- T1 Circuitsdard PCM encoding(G.711) scores 4.1, and On a Tl circuit, there are up to 24 channels available for voice. 64k from conversation 1 is loaded into the first TlG.729 scores 3.92. channel, then 64k from the conversation 2 is loaded into the second channel, and so on. If not enough conversations exist to fill the available channels, they are padded with null values. The 24 channels are grouped together as a frame. Depending on the implementation, either 12 frames are grouped together as a larger frame (called SuperFrame or SF), or © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  35. 35. 24 frames are grouped together (called Extended SuperFrame or ESF). T l s are typically full duplex, with two wiressending and the other two wires receiving.E1 CircuitsAn El is very similar to a T l . There are 32 channels, of which 30 can be used for voice. (The other two are used forframing and signaling, respectively.) The 32 channels are grouped together as a frame, and 16 frames are groupedtogether as a multiframe. El circuits are common in Europe and Mexico, with some El services becoming available inthe United States.Channel Associated Signaling (CAS)—T1Although the 64 k channels of a Tl are intended to carry digitized voice, we must also be able to transmit signaling infor-mation, such as on-hook and off-hook, addressing, and so forth. In CAS circuits, the least significant bit of each channelin every sixth frame is "stolen" to generate signaling bit strings. SF implementation takes 12 frames and creates aSuperFrame. Using one bit per channel in every sixth frame gives two 12-bit signaling strings (known as A and B) perSuperFrame. The A and B strings are used to signal basic status, addressing, and supervisory messages. In ESF, 24 chan-nels are in an Extended SuperFrame, which gives A, B, C, and D signaling strings. These can be used to signal moreadvanced supervisory functions.Because CAS takes one bit from each channel in every sixth frame, it is known as Robbed Bit Signaling (RBS). UsingRBS means that a slight degradation occurs in voice quality because every sixth frame has only 7 instead of 8 bits torepresent the sample; however, this is not generally a perceptible degradation. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  36. 36. Channel Associated Signaling (CAS)—T1El signaling is slightly different. In an El CAS circuit, the first channel (channel 0 or timeslot 1) is reserved for framinginformation. The 17th channel (channel 16 or timeslot 17) contains signaling information—no bits are robbed from theindividual channels. Timeslots 2 - 1 6 and 18-32 carry the voice data. Each channel has specific bits in timeslot 17 forsignaling. This means that although El CAS does not use RBS, it is still considered C A S ; however, the signaling is out-of-band in its own timeslot.Common Channel Signaling (CCS)CCS provides for a completely out-of-band signaling channel. This is the function of the D channel in an ISDN PRI orBRI implementation. The full 64 k of bandwidth per channel is available for voice; instead of generating A B C D bits, aprotocol known as Q.931 is used out-of-band in a separate channel for signaling. An ISDN PRI Tl provides 23 voicechannels of 64 k each (called Bearer or B channels) and one 64 k D (for Data) channel (timeslot 24) for signaling. AnISDN PRI El provides 30 B channels and 1 D channel (timeslot 17); an ISDN BRI circuit provides two 64 k B channelsand one D channel of 16 k. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  37. 37. U n d e r s t a n d i n g VoIPThe elements of traditional telephony—status, address and supervisory signaling, digitization, and so on—must havefunctional parallels in the VoIP world for systems to function as people expect them to, and more importantly, for VoIP tointeract with the PSTN properly.This section examines packetizing digital voice, signaling, and transport protocols, the components of a VoIP network,and the factors that can cause problems in VoIP networks and how they can be mitigated.Understanding PacketizationIP networks move data around in small pieces known as packets. Because we know how to digitize our voice, it nowbecomes just another binary payload to move around in a packet. VoIP uses Digital Signal Processors (DSP) for the codecfunctions. The digitized voice is then packaged in an appropriate protocol structure to move it through the IP infrastructure.DSPsDSPs are specialized chips that perform high-speed codec functions. DSPs are found in the IP phones to encode theanalog speech of the user and to decode the digitized contents of the packets arriving from the other end of the call. DSPsare also used on IOS gateways at the interface to PSTN circuits, to change from a digital circuit to packetized voice, orfrom an analog circuit to packetized voice. DSPs also change from one codec to another, allow conferencing and callpark, and other telephony features. DSPs are a vital component of a VoIP system. Different chip types have varyingcapacities, but the general rule is that you want as many D S P resources available to you as possible. The D S P calculatoron will help you calculate what you must have.Real-Time Transport Protocol (RTP)RTP was developed to better serve real-time traffic such as voice and video. Voice payloads are encapsulated by RTP, thenby UDP, then by IP. A Layer 2 header of the correct format is applied; the type obviously depends on the link technologyin use by each router interface. A single voice call generates two one-way RTP/UDP/IP packet streams. U D P providesmultiplexing and checksum capability; RTP provides payload identification, timestamps, and sequence numbering. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  38. 38. Payload identification allows us to treat voice traffic differently from video, for example, simply by looking for the RTPheader label, simplifying our configuration tasks. Timestamping and sequence numbering allows VoIP devices to reorderRTP packets that arrived out of sequence and play them back with the same timing in which they were recorded, elimi-nating delays or jerkiness. There is no provision for retransmission of a lost RTP packet.Each RTP stream is accompanied by a Real-Time Transport Control Protocol (RTCP) stream. RTCP monitors the qualityof the RTP stream, allowing devices to record events such as packet count, delay, loss, and jitter (delay variation).A single voice packet by default contains a payload of 20 msec of voice (either uncompressed or compressed). Becausesampling is occurring at 8000 times per second, 20 msec gives us 160 samples. If we divide 8000 by 160, we see that weare generating 50 packets with 160 bytes of payload, per second, for a one-way voice stream.If we use compression, we can squeeze the 160-byte payload down to 20 bytes using the G.729 codec. We still have 160samples, still 20 msec of audio, but reduced payload size.CodecsThe codecs supported by Cisco include the following: • G.711 (64kbps)—Toll-quality voice, uncompressed. • G.729 (8kbps) • Annex A variant: less processor-intensive, allows more voice channels encoded per DSP chip; lower audio quality than G.729 • Annex B variant: Allows the use of Voice Activity Detection and Comfort Noise Generation; can be applied to G.729 or G.729-AThe values for bandwidth shown do not include the Layer 3 and Layer 2 overhead; the actual bandwidth used by a single(one-way) voice stream can be significantly larger. The following tables summarize the additional overhead added bypacketization and Layer 2 encapsulation (assume 50 packets per second (pps): © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  39. 39. Bandwidth Calculation, Without Layer 2Codec G.711 G.729Voice Payload 160 Bytes 20 BytesRTP Header 12 Bytes 12 BytesUDP Header 8 Bytes 8 BytesIP Header 20 Bytes 20 BytesTotal Before Layer 2 200 Bytes 60 BytesTotal Bitrate @ 50 pps 80,000 bps (80 kbps) 24,000 bps (24 kbps)Bandwidth Calculation, With Layer 2Layer 2 Type G.711 = 2 0 0 Bytes/packet G.729 = 60 Bytes/packetEthernet 18 Bytes 18 BytesMultilink PPP 6 Bytes 6 BytesFrame Relay FRF. 12 6 Bytes 6 BytesTotal incl. Layer 2 218 Bytes 206 Bytes 206 Bytes 78 Bytes 66 Bytes 66 BytesTotal Bitrate incl. Layer 2 87.200 82,400 82,400 31,200 26,400 26,400(@ 50 pps) (87.2 kbps) (82.4 kbps) (82.4 kbps) (31.2 kbps) (26.4 kbps) (26.4 kbps)When using G.729, the RTP/UDP/IP header of 40 bytes is twice the size of the 20B voice payload. This consumes signif-icant bandwidth just for header transmission on a slow link. The recommended solution is to use Compressed RTP(cRTP) on slow WAN links. cRTP reduces the RTP/UDP/IP header to 2 bytes without checksums or 4 bytes with check-sums. The effect of using cRTP is illustrated in the following table. (Note: Ethernet is not included because it is not clas-sified as a slow link.) © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  40. 40. Bandwidth Calculation, Using cRTPCodec G.711 G.729Voice Payload 160 Bytes 20 BytescRTP header w/ chksum 4 Bytes 4 BytescRTP header no chksum 2 Bytes 2 BytesTotal before Layer 2: 164 Bytes 162 Bytes 24 Bytes 22 BytesMultilink PPP orFrame Relay FRF. 12 6 Bytes 6 BytesTotal WAN bandwidth @50 pps incl. Layer 2: 68000 bps 67,200 bps 12,000 bps 11,200 (68 kbps) (67.2 kbps) (12 kbps) (11.2 kbps)Voice Activity Detection (VAD)Phone conversations on average include about 3 5 % silence. In Cisco Unified Communications, by default silence is pack-etized and transmitted, consuming the same bandwidth as speech. In situations where bandwidth is very scarce, the VADfeature can be enabled, causing the voice stream to be stopped during periods of silence. The theory here is that the band-width otherwise used for silence can be reclaimed for voice or data transmission. VAD also adds Comfort NoiseGeneration (CNG), which fills in the dead silence created by the stopped voice flow with white noise. VAD should not betaken into account during the network design bandwidth allocation process because its effectiveness varies with back-ground noise and speech patterns. VAD is also made ineffective by Music on Hold and fax features. In reality, VAD typi-cally causes more problems than it solves, and it is usually wiser to add the necessary bandwidth. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  41. 41. Additional DSP FunctionsIn addition to digitizing voice, D S P resources are used for the following: • Conferencing: DSPs mix the audio streams from the conference participants and transmit the mix (minus their own) to each participant. • Transcoding and Media Termination Points (MTP): A transcoder changes a packetized audio stream from one codec to another, perhaps for transit across a slow WAN link. MTPs provide a point for the stream to be terminated while other services are set up. • Echo Cancellation: DSPs provide the calculation power needed to analyze the audio stream and filter out the repeti- tive patterns that indicate echo. Echo is a chief cause of perceived poor voice quality; echo cancellation is an impor- tant function. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  42. 42. I n t r o d u c i n g VoIP Signaling ProtocolsVoIP signaling protocols are responsible for call setup, maintenance, and teardown. A number of different protocols are inuse—some standards-based, others proprietary, and each with advantages and disadvantages. The following sectionsintroduce the signaling protocols you should know about, including SCCP, H.323, MGCP, and SIP.VoIP Signaling ProtocolsVoIP signaling protocols handle the call setup, maintenance, and teardown functions of VoIP calls. It is important to keepin mind that the signaling functions are an entirely separate packet stream from the actual voice bearer path (RTP). Thesignaling protocol in use must pass the supervisory, informational, and address information expected in any telephonysystem.VoIP signaling protocols are either peer-to-peer or client-server; in the case of peer-to-peer protocols, the endpoints havethe intelligence to perform the call-control signaling themselves, Client-server protocols send event notifications to thecall agent (the Unified CM server) and receive instructions on what actions to perform in response. The following tablesummarizes the characteristics of the four signaling protocols dealt with here. Inter-Vendor Implemented ImplementedProtocol Standard? Compatibility on Gateways on Cisco IP Phones Operating ModeH.323 Yes--ITU Very Good Yes No Peer-to-PeerMGCP Yes--IETF Good Yes No Client/ServerSIP Yes--IETF Basic Yes Yes; also third-party phones Peer-to-PeerSCCP N o - -Cisco Proprietary Cisco only Some Cisco IP Phones only Client/ServerH.323H.323 is not itself a protocol; it is an umbrella standard that defines several other related protocols for specific tasks.Originally conceived as a multimedia signaling protocol to emulate traditional telephony functionality in IP L A N © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  43. 43. environments, it is a long-established and stable protocol very suitable for intervendor compatibility. H.323 is supportedby all Cisco voice gateways and CM platforms as well as some third-party video endpoints.MGCPMedia Gateway Control Protocol is a lightweight client/server protocol for PSTN gateways and some clients. It is simpleto configure and allows the call agent to control the M G C P gateway, eliminating the need for expensive gateways withintelligence and complex configurations. The gateway reports events such as a trunk going off-hook, and the call agentinstructs the gateway on what to do; the gateway has no local dial plan because all call routing decisions are made at thecall agent and relayed to the M G C P gateway. M G C P is not as widely implemented as SIP or H.323. M G C P is notsupported by Unified CM Express or the Smart Business Communication System.SIPSession Initiation Protocol is an IETF standard that uses peer-to-peer signaling. It is very similar in structure and syntaxto HTTP, and because it is text-based, it is relatively simple to debug and troubleshoot. SIP can use multiple transportlayer protocols and can support security and proxy functions. SIP is an evolving standard that currently provides basictelephony functionality; further developments and extensions to the standard will soon make it feature-comparable withSCCP. One of its most important capabilities is creating SIP trunks to IP Telephony service providers, replacing orenhancing traditional T D M PSTN connections.SCCPSkinny Client Control Protocol is a Cisco-proprietary signaling protocol used in a client-server manner between UnifiedCM and Cisco IP Phones (and some Cisco gateways). SCCP uses TCP connections to the Unified CM to set up, maintain,and tear down voice and video calls. It is referred to as a stimulus protocol, meaning that it sends messages in response toevents such as a phone going off-hook or a digit being dialed. SCCP is the default signaling protocol for all Cisco IPphones, although many also support SIP; SIP does not yet support the full feature set available to SCCP phones. AllCisco Unified Communications call agents (CM, CM Express, and the 500 Series) and some gateways support SCCP. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  44. 44. C o n n e c t i n g a VoIP S y s t e m to a S e r v i c e ProviderNetworkA VoIP system that can place calls only to other devices on the IP network is only marginally useful; we still need toplace calls out to the PSTN, and to do so we need to connect to a phone service provider, whether via traditional T D Mlinks or ITSP connections. The device that acts as the interface to the PSTN is the voice gateway; it provides the physicalconnection and logical translation between two or more different network technologies.Understanding Gateways, Voice Ports, and Dial PeersThe following sections establish some terms of reference.GatewaysIn the Cisco Unified Communications architecture, a gateway is typically a voice-enabled router with an appropriatevoice port installed and configured. Gateways can have both analog and digital voice port connections, including analogFXO, F X S , and E & M or digital T l / E l or PRI interfaces.Call LegsA call leg is the inbound or outbound call path as it passes through the gateway. As the call comes into the gateway, it isassociated with an inbound port. (This is the inbound call leg.) As the call is sent out another gateway port, this createsthe outbound call leg. There will be inbound and outbound call legs at each gateway router.Dial PeersA dial peer is a pointer to an endpoint, identified by an address (a pattern of digits). Cisco gateways support two types ofdial peers: POTS and VoIP. POTS dial peers are addressed with PSTN phone numbers, and VoIP dial peers are addressedby IP addresses. Dial peers identify the source and destination endpoints of call legs; an inbound call leg is matched to adial peer, and the outbound call leg is routed to a destination dial peer. Depending on the direction of the call, the dial peersmay be POTS inbound and VoIP outbound, vice versa, or possibly both VoIP. It is unlikely but not impossible that the © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  45. 45. inbound and outbound dial peers would both be POTS. Each dial peer also defines attributes such as the codec to use, QoSsettings, and other feature settings. Dial peers are configured in the gateway IOS, using either the CLI or GUI interface.The partial output that follows shows a simple POTS dial peer configuration:G a t e w a y ( c o n f i g ) # d i a l - p e e r v o i c e 10 p o t sGateway(config-dialpeer)#destination-pattern 8675309Gateway(config-dialpeer)#port 1/0/1The number assigned to dial peers is arbitrary, although dial peer 0 exists by default and cannot be deleted. The keywordpots creates a POTS dial peer; the keyword voip would create a VoIP dial peer. The destination-pattern command iden-tifies that the attached device (phone or PBX) terminates calls to the specified number (or a range of numbers if connect-ing to a PBX). The port command identifies the physical hardware connection the dial peer will use to reach thedestination pattern.The destination-pattern command associates a phone number with a dial peer. The specified pattern can be a specificphone number (as above, 8675309) or an expression that defines a range of numbers. The router uses the patterns todecide which dial peer (and associated physical port) it should route a call to. The following table briefly explainsdestination-pattern syntax.Character Meaning+ The preceding digit is repeated one or more times.* and # NOT wildcards; these are valid DTMF digits., (comma) Inserts a one-second pause.. (dot) Specifies any one wildcard digit (0 - 9, *, #). The pattern "20." would match all strings from 200 through 209, plus 20*and20#.[] Square brackets define a range, within which any one digit may be matched; for example, "20[4-6]" defines 204, 205, and 206.T Indicates a variable length string; this is useful in cases where local, long-distance, and international PSTN numbers may be called; the destination pattern could men be ".T". This pattern would match any string of up to 32 digits. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  46. 46. C o n n e c t i n g a VoIP S y s t e m to a S e r v i c e P r o v i d e r N e t w o r k Configuring VoIP dial peers is equally simple. Examine the following configuration: G a t e w a y ( c o n f i g ) # d i a l - p e e r voice 20 voip Gateway(config-dialpeer)# destination-pattern 4.... Gateway(config-dialpeer)# session target ipv4: In this example, the destination pattern is any four-digit number starting with "4." A new command, session-target, is used to identify the IP (version 4 in this case) address of the gateway or call agent that will terminate the call. If the IP address is on a router, it should be a loopback IP so that the address is always available even if a physical interface fails. The preceding configuration creates an H.323 dial peer (in contrast to a SIP dial-peer). Routers attempt to match dial peers for the inbound call leg according to the following rules:NOTE 1. Look for the incoming called-number command in a dial peer that matches the called number or DNIS string of theThe default dial peer 0 inbound leg.cannot be deleted ormodified. It does not 2. Look for the answer-address command in a dial peer that matches the calling number or ANI string of the inboundnegotiate services such call VAD, DTMF Relay, or 3. Look for the destination-pattern command in a dial peer that matches the calling number or ANI string of theTCL applications. Thedial peer 0 configuration inbound call leg.for inbound VoIP calls 4. Look for the POTS dial peer port command that matches the voice port of the incoming call (POTS dial peers only).contains the following:• Any codec 5. If all of the above fail to match, match against Default Dial Peer 0 as a last resort.• VAD enabled The default dial peer 0 config for inbound POTS calls includes the following:• No RSVP Support • no ivr application• Fax-rate voice When a router is matching the dialed digits against the patterns in the configured dial peers, it attempts to find the longest match. This occurs on a digit-by-digit basis. Each successive digit may validate some patterns while eliminating others until a single pattern represents the longest match between the dialed digits and the destination pattern, at which point the call is routed to the outbound dial peer configured with that matching pattern. © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  47. 47. Consider the following configuration:d i a l - p e e r voice 10 voipdestination-pattern .Tsession t a r g e t ipv4: i a l peer v o i c e 2 0 v o i pdestination-pattern 867[2-3]...session t a r g e t ipv4:!d i a l - p e e r voice 30 voipdestination-pattern 8674...session t a r g e t i p v 4 : i a l - p e e r voice 40 voipd e s t i n a t i o n - p a t t e r n 8675309session t a r g e t ipv4: this configuration, the following example dialed numbers illustrate how the patterns match dialed digits: • The dialed number 867-5309 will match dial peer 40 (exact 7-digit match) • The dialed number 867-4309 will match dial peer 30 (first four digits match) • The dialed number 867-3309 will match dial peer 20 (first four digits match) • The dialed number 876-5309 will match dial peer 10 (no other exact match, so the " . T " pattern matches)Internet Telephony Service ProvidersAs VoIP technology matured and stabilized, telephone service providers began extending VoIP connectivity to theircustomers, allowing for simple, flexible connection alternatives to traditional T D M links. Internet Telephony Service © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  48. 48. Providers (ITSP) connections are typically much less expensive, available in smaller bandwidth increments than Tl orPRI links, and can route nonvoice data traffic concurrently. QoS configuration is supported (and in fact is required forproper VoIP operation). Most ITSP links use SIP, but H.323 is an option. The gateway configuration is relatively simple,with the creation of a VoIP dial peer pointing at the provider with the parameters they supply. PSTN calls are routed tothe provider, who then routes calls to their PSTN connection, usually with a toll-minimizing route that dramaticallyreduces long-distance costs to the customer.Understanding Call Setup and Digit ManipulationSuccessfully completing a phone call requires that the correct digits are sent to the terminating device, whether on theVoIP network or the PSTN. PSTN calls are typically more complex because of the varying local and internationalrequirements for the number of digits required to route the call. Over and above this basic requirement are the additionalcomplexities imposed by requirements of the business: we may want to change our A N I number, add or strip accesscodes, compensate for undesirable default behavior, or build specialized functionality for our particular purposes. Thissection deals with digit manipulation and troubleshooting.Digit Consumption and ForwardingSome strange things happen when an IOS gateway matches a dial peer for an outbound call leg and forwards the dialeddigits to the terminating device.For POTS dial peers, the gateway consumes (meaning strips away) the left-justified digits that exactly match the dial-peerdestination pattern and forwards only the wildcard-matched digits to the terminating device. Clearly, this could causeproblems if the PSTN were to receive only 4 digits, as in this example:d i a l - p e e r voice 20 potsdestination-pattern 867....port 1/0:1 © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  49. 49. With this configuration, if the dialed number was 867-5309, the gateway would forward only 5309 (the wildcardmatches), and the PSTN would be unable to route the call. Adding the command no digit-strip in the dial-peer configura-tion will change this behavior and cause the gateway to forward all dialed digits.For VoIP dial peers, the default behavior is to forward all collected digits.Digit CollectionThe router will collect digits one at a time and attempt to match a destination pattern. As soon as it has an exact match,the call is immediately placed, and no more digits are collected. If there are destination patterns that have overlappingdigits, this can cause calls to be misrouted, as in the following example:Dial-peer voice 1 voipD e s t i n a t i o n p a t t e r n 555Session t a r g e t ipv4:!Dial-peer voice 2 voipD e s t i n a t i o n - p a t t e r n 5552112Session t a r g e t ipv4: the user dials 555-2112, dial peer 1 will exactly match at the third digit, the call will be immediately routed using dialpeer 1, and only the collected digits of 555 will be forwarded. We solve the problem by changing the configuration asshown next:Dial-peer voice 1 voipDestination pattern 5 5 5 . . . .Session t a r g e t ipv4:!Dial-peer voice 2 voipD e s t i n a t i o n - p a t t e r n 5552112Session t a r g e t ipv4: © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
  50. 50. Now, when the third digit is entered, the router cannot make an exact match because both dial peers are possible matches;when the last digit is dialed, the router determines that dial peer 2 is an exact match and immediately places the call. Dialpeer 1 is also a match, but because of the wildcards, the destination pattern matches 10,000 possible numbers (0000through 9999); it is not as close a match as dial peer 2.Digit ManipulationSometimes we need to add, change, or remove digits before the call is placed. We do this to avoid inconveniencing usersor to match the dialed digit requirements of a gateway or the PSTN. We have several methods of modifying the digitstring, as described in the following sections.prefixThe prefix dial-peer command adds digits to the beginning of the string after the outbound dial peer is matched butbefore passing digits to the destination. An example of its use is a POTS dial peer with 2 . . . as the destination pattern. Ifthe user dials 2112, the default behavior is for the POTS dial peer to forward only 112. Adding the command prefix6045552 forces the router to prepend the additional digits required to route the call over the PSTN:d i a l - p e e r voice 20 potsdestination-pattern 2...prefix 6045552port 1/0/0forward-digitsforward-digits: This dial-peer command forces the specified number of digits to be forwarded, whether the digits wereexact match or wildcard matches, overriding the default behavior of stripping the exact matches. You can specify anumber of digits to forward (as shown in the example that follows) or use forward-digits all to force all dialed digits tobe forwarded.d i a l - p e e r voice 20 potsdestination-pattern 5552... © 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.