CCNA Voice Quick Reference Michael Valentine
ciscopress.com
Your Short Cut to Knowledge
As a final exam preparation tool, the CCNA Voice Quick Reference provides a concise review of all objectives on the new IIUC exam (640-460). This digital Short Cut provides you with detailed, graphical-based information, highlighting only the ke y topics in cram-style format. With this document as your guide, you will review topics on concepts and commands that apply to Cisco Unified Communications for small and medium-sized businesses. This fact-filled Quick Reference allows you to get all-important information at a glance, helping you focus your study on areas of weakness and enhancing your memory retention of essential exam concepts.
About the Author Mike Valentine has 13 years of experience in the IT field, specializing in network design and installation. He is currently a
Cisco trainer with Skyline Advanced Technology Services and specializes in Cisco Unified Communications, CCNA, and CCNP classes. His accessible, humorous, and effective teaching style has demystified Cisco for hundreds of students since he began teaching in 2002. Mike holds a bachelor of arts degree from the University of British Columbia and currently holds the MCSE: Security, CCNA, CCDA, CCNP, CCVP, IPTX, QoS, CCSI #31461, CIEH, and CTP certifications. He has completed the CCIE written exam. Mike was on the development team for the Cisco Unified Communications Architecture and Design official Cisco coursewa re and is currently developing custom Unified Communications courseware for Skyline. Mike coauthored the popular CCNA Exam Cram, second edition, first published in December 2005, as well as the third edition of that volume published in
December 2007.
A b o u t t h e T e c h n i c a l E d i t or Denise Donohue, CCIE No. 9566, is manager of Solutions Engineering for ePlus Technology in Maryland. She is responsible
for designing and implementing data and VoIP networks and supporting companies based in the National Capital region. Pr ior to this role, she was a systems engineer for the data consulting arm of SBC/AT&T. Denise was a Cisco instructor and cours e director for Global Knowledge and did network consulting for many years. © 2 008 Cisco Systems Inc. Inc. All rights reserved. This publication publication is protected by copyright. Please see page 147 for more details.
As a final exam preparation tool, the CCNA Voice Quick Reference provides a concise review of all objectives on the new IIUC exam (640-460). This digital Short Cut provides you with detailed, graphical-based information, highlighting only the ke y topics in cram-style format. With this document as your guide, you will review topics on concepts and commands that apply to Cisco Unified Communications for small and medium-sized businesses. This fact-filled Quick Reference allows you to get all-important information at a glance, helping you focus your study on areas of weakness and enhancing your memory retention of essential exam concepts.
About the Author Mike Valentine has 13 years of experience in the IT field, specializing in network design and installation. He is currently a
Cisco trainer with Skyline Advanced Technology Services and specializes in Cisco Unified Communications, CCNA, and CCNP classes. His accessible, humorous, and effective teaching style has demystified Cisco for hundreds of students since he began teaching in 2002. Mike holds a bachelor of arts degree from the University of British Columbia and currently holds the MCSE: Security, CCNA, CCDA, CCNP, CCVP, IPTX, QoS, CCSI #31461, CIEH, and CTP certifications. He has completed the CCIE written exam. Mike was on the development team for the Cisco Unified Communications Architecture and Design official Cisco coursewa re and is currently developing custom Unified Communications courseware for Skyline. Mike coauthored the popular CCNA Exam Cram, second edition, first published in December 2005, as well as the third edition of that volume published in
December 2007.
A b o u t t h e T e c h n i c a l E d i t or Denise Donohue, CCIE No. 9566, is manager of Solutions Engineering for ePlus Technology in Maryland. She is responsible
for designing and implementing data and VoIP networks and supporting companies based in the National Capital region. Pr ior to this role, she was a systems engineer for the data consulting arm of SBC/AT&T. Denise was a Cisco instructor and cours e director for Global Knowledge and did network consulting for many years. © 2 008 Cisco Systems Inc. Inc. All rights reserved. This publication publication is protected by copyright. Please see page 147 for more details.
C C N A Voice Qui ck Refer ence
by Michael Valentine Valentine
Introduction
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 who m are not even even aware they are are usi ng it. Ci sco has aggressivel y pursued the development and deployment of its Unifi ed Communi cation s suite of products products and can now offer an integrated integrated voice, video, and data solution for any busines s, whether it has has just a few employees or a hundred thousand worldwid e. The technology is reliable, user friendly, and exciting, but it is not simp le—and a successful deployment requires that that the the designers, implementers, and administra tors of a Unified Communications system know what they are doing. Trainin g and certification certification of key staff are are strategic strategic components of any any business plan to deploy a Un ifie d Communi catio ns system. Unt il recently, recently, the the training track track for Unifie d Communicat ions went from the C C N A (the (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 VoIP signaling protocols, Unified Communications Manager administration, traditional telephony, telephony, gateway and gatekeep gatekeeper er configuration, Qo S , and so on—all the while assumi ng that the the student student had a firm grasp of routing and switching concepts. I have met many good C C N A students students who had no telephony telephony or Vo IP back ground and consequently had great great difficulty in the C C V P program. Likewise, many students with very strong traditional telephony telephony experience experience were quickl y lost in the intensive intensive data concepts concepts of the C C V P curricul um. 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 certificatio certifications ns at the the Associate level. The I I U C curric ulum prepares prepares student students s for the the C C N A Voice certification, certification, which in turn is a solid preparation for and a much-needed transition to C C V P .
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CCNA Voice Quick Reference
by Michael Valentine
Introduction
Purpose of This Guide 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.
Who Should Read This Guide 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 CCVP exam may turn to this guide as a refresher, too. Then there are alway s 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 Today's 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.
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CCNA Voice Quick Reference
by Michael Valentine
Introduction 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 mainta ining the network sim pler and more cost effective than if three separate systems existed. Unified Com munic ation s 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 component's 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.
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FIGURE
1
The Unified Communications Architecture
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•
•
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 Layer At the infrastructure layer, we are building the connections between all the devices that send and receive data, voice, and video. These include Layer 2 and 3 switches, routers, and voice gateways. Voice gateways are among the most important components because they provide the connection to the PSTN or other network carriers. One of the critical functions (and one that is unfortunately often underemphasized in many deployments) is quality of service, or QoS. QoS provides service 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;
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if any weaknesses exist here, they will manifest as system failures or unreliability. It is very important to build a solid and correct foundation. The goal is to achieve 99.999% uptime; achieving that goal takes careful attention and good design.
Call Processing Layer The call processing layer is chiefly about the call agents. A call agent is not a person; it is an application that looks at the signaling traffic from devices that place and receive calls, and it determines what to do with the call. A Unified IP Phone sends a packet to the call agent when you lift the receiver; the call agent instructs the phone to play a dial tone. When you begin dialing a number to call, the call agent receives the digits and tries to find a match for the number in its dial plan. If the destination number is a phone that it controls, it tells the called device to ring. During the call, the call agent also sets up other services, such as Hold, Call Park, Transfer, Conference, and so on. The call agent also instructs the phones to tear down the call when one party hangs up. The call agent usually keeps detailed records of each call made; these are commonly 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.
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Smart Business Communications System FIGURE 2 The Smart Business Communications System—Image © Cisco
The Smart Business Communications System is a group of specially designed, integrated devices that can provide highquality 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.
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Unified Communication Manager Express FIGURE 3 Cisco Integrated Services Routers for Unified Communications Manager 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 fullfeatured 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 MCS 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
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Reach), Do Not Disturb, Intercom Whisper, and Audible Message Waiting Indication, as well as speech recognition and integrated messaging. Because Unified CM Business Edition uses the same call agent software as a full cluster deploy ment of Unified CM, it supp orts full integration with the other Unified Comm unic ation s applicatio ns, su ch as Unified Presence, Unified Personal Communicator, MeetingPlace Express, Contact Center Express, and so on.
Unified Communications Manager The full version of Unified Commun icati ons Mana ger is an enterprise-c lass, fully scalable, redundan t, 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 CM versions prior to 5.x are Windows based, whereas versions 5.x and 6.x are Linux-based appliances.
Applications Layer There are effectively a limitless number of applications that can be part of a Unified Communications system, because third-party applications can be developed to closely integrate with the Cisco suite of products. The following is a list of the 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 911, 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.
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•
•
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 Pres ence : 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 Layer An increasing variety of Cisco Unified IP Phones (and third-party IP phones) can be part of a Unified Communications deployment. All Cisco Unified IP Phones provide a display-based user interface, user customization, Power over Ethernet capability (where appropriate), and support for G.711 and G.729 codecs (and, on some models, Cisco Wideband and/or iLBC 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 life
Mobility 7921G: Wireless, 2-in. color screen, speakerphone, XML-PTT, longer battery life IP Communicator: Software-based IP Phone, emulates 7970G functionality
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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-capable
Advanced
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-capable
Color
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-capable
Video 7985G: Personal desktop video phone Unified Video Advantage: Software IP Video Phone with support for attached camera
Conference 7936G: Backlit LCD, 3 softkeys, small-medium conference needs 7937G: Hi-fidelity audio, extended audio coverage w/ extra mics, large display © 20 08 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
Understanding Unified Communications Applications In 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.
Messaging A variety of messag ing optio ns are available to suit the needs of businesse s small and large. T he following table provid es a summary of the options. Product
Max. Users
Messaging Capability
Platform
TDM PBX Integration?
Networking?
Redundancy?
Unity Express
250
Voice Mail + Integrated Messagin g
ISR
No
Yes
No
Unity Connection
3000
Voice Mail + Integrated Messaging
MCS
Yes
No
No
Unity
7500 per server
Voice Mail + Integrated Messaging + Unified Messaging
MCS
Yes
Yes
Yes
The following sections describe the messaging products listed in the table in more detail.
Cisco Unity Express Unity Express is an ISR-based application that runs either on an AIM module or an NM module. AIM modules are connected to the main board as a daughter board addition and use flash memory for greetings and message storage. AIM modules therefore have less capacity for storage. NM modules are inserted into module bays in ISR routers, use a hard
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disk for greeting and message storage, and have greater capacity for storage than AIM modules. Unity Express supports from 4 to 16 concurrent sessions and 12 to 250 mailboxes (dependent on the module and platform installed). Unity Express is managed through the command line or a web-based GUI. It allows users to view and sort their voice messages using the IP Phone display, email application, or IMAP client. Unity Express can be deployed in conjunction with Unified CM or CM Express and can supplement a full Unity deployment.
Cisco Unity Connection Unity Conn ection is a med ium-si ze business solutio n with a full range of messag ing features. It can be deploy ed on its own 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 a standalone application, Unity connection supports up to 3000 users per server (dependent on hardware). Scalability is achieved by networking up to 10 other Unity messaging products of any type. Fourteen languages are supported for deployments worldwide. Unity Connection also supports speech recognition, allowing users to speak commands to manage their messages hands-free. Multiple interfaces are supported for managing messages from an IP Phone, an email client, a web GUI, or Cisco Unified Personal Communicator. Users can define their own rules to transfer calls based on caller, time of day, and Microsoft Exchange calendar status.
Cisco Unity Unity is the enterprise-class messaging application with support for up to 7500 users per server and up to 250,000 users in a multi server netwo rked environ ment. Interop erability with legacy voice-ma il systems, notably Octel and Nortel systems, 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 their emails read to them over the phone by the RealSpeech engine; speech recognition is also available so users can instruct Unity to play, search, or record messages hands-free. Secure messaging is supported, allowing encrypted messages and preventing messages that have expired from being played. Access to messages is made simple, intuitive, and possible from almost anywhere.
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Auto Attendant An 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 response to the greeting they heard. If you have ever heard: "For service in English, press I. Pour service en Francais, appuye z sur le 2...," you have been served by an Auto Attendant. Typically, Auto Attendants allow callers to select the department or extension they want to call, and often they allow the caller to spell out a first or last name to search in the company di rec tory. Cisco Unity, Unity Connection, and Unity Express all provide Auto Attendant functionality; Unity and Unity Connection include a simple web interface that makes it very easy to construct menus and test to see that they work as you intended.
Cisco Unified IP IVR Although 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-digit account number, followed by the # sign"), Text-to-Speech, database integration, and Java application integration, a much more advanced IVR application is required. Cisco Unified IP IVR has all these advanced capabilities. Call centers that have a high call volume and many possible queues of callers waiting for different agent capabilities can effectively deploy Unified IP IVR to steer callers to the correct agent, or perhaps to an automated information source without the need to t ie up an agent at all. Unified IP IVR includes the capability to provide both real-time and historical reports on its utilization and offers multiple-language support.
Cisco Unified Customer Voice Portal For 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.
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Cisco Unified Contact Center Cisco provides a range of Contact Center products for SMB, Enterprise, and Service Provider applications. Customer contact solutions provide multiple avenues to reach and interact with customers, including basic telephony as well as feature-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 singleserver, 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 agent's 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 Solutions Today's 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 features in 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 worker's enterprise
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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 user's 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.
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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 nearzero latency. This can only be described as a high-end solution, with commensurate demands on bandwidth. FIGURE 4 The Cisco Telepresence 3000 System—Image © Cisco
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Understanding Traditional Telephony This section introduces traditional telephony systems, concepts, and applications.
The PSTN Public Switched Telephone Network
FIGURE 5
A Representation of the Public Switched Telephone Network (PSTN)
The PS TN, or Public Sw itched Telepho ne Network, is made up of Centra l 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.
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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 Businesse s have more elaborate require ments of the telephon e bey ond 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 Systems FIGURE 6 A Representation of a PBX 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 5digit 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 u se, a switching engine that determines which port to route a call out, line cards to connect to phones, and trunk cards to © 2 008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
CCNA Voice Quick Reference
by Michael Valentine
Understanding 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 don't 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: •
O n - H o o k : 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
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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 someo ne picks up the pho ne and taps the receiver cradle repeated ly; 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. SS7's 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. FIGURE 7 PSTN Call Setup
0 Customer Telephone
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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 Plans NOTE
A numbering plan is an organized distribution of telephone numbers administered by a regional or national authority. The
Codes do not always need to be dialed; Local numbers must always be dialed.
plan defines the rules that allocate numbers according to an established international telecommunications standard. For example, the North American Numbering Plan defines a country code of 1, followed by a three-digit area code, a threedigit office code, and a four-digit local number. There are numerous other numbering plans for other countries or regions of the world.
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The North American Numbering Plan Let's look at the NANP 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-XXXX NOTE Several other ranges are reserved for specialized purposes. One commonly recognized one is 55501XX, which is used in film and TV, demonstra tions, or education. No actual customer is assigned these numbers, so calling a number seen in a movie will not pose a nuisance to anyone. When Tommy Tutone recorded "867-5309/Jenny," he immediately annoyed thousands of phone customers worldwide.
It is very important to note that the "N" represents any digit in the range 2 through 9, and the "X" represents any digit 0 through 9. You will never find an office or area code of OXX or 1XX; those numbers are either reserved for specialized purposes or would interfere with things like operator access numbers. Several ranges are also reserved for Easily Recognizable Codes (ERCs); these are numbers where the second and third numbers of the area code are the same. The y are used for special services—for example, 800, 888, 877, and 866 are toll-free numbers. Another recognizable assign ment is the " N i l " series: this includes 41 1, 6 11 , and 911 numb ers that are not used as area codes but for other special assignments, such as information or emergency services.
E.164 Addressing The E.164 addressing scheme is an international standard for telephone numbering plans, originally developed by the International Telecommunication Union. An E.164 number contains the following components: CC—Country Code NDC—National Designation Code SN—Subscriber Number An E.164 number is standardized at 15 digits, generating over 100 trillion unique strings. In theory, it's possible to direct dial any conventional phone in the world from any other conventional phone.
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Introduction to Analog Circuits Analog (in contrast to digital) circuits are still the most common telephone connections worldwide. The phone line to a North American home is most commonly an analog loop circuit, although more and more digital phone services are being installed. Cisco gateways must connect to various analog services to place calls to the PSTN; the analog circuits that Cisco 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 Phone An 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 call
Foreign Exchange Station An FXS port connects directly to an analog phone or fax machine. Switches (including CO switches and PBXs) and Cisco 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 VoIP network 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 analog station to the VoIP network. © 20 08 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
Foreign Exchange Office An FXO 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 FXO ports. These ports allow the gateway to place and receive calls to/from the PSTN. FXO ports also include a codec.
FIGURE 8 Loop-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!
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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 beca use of the open circuit.
2. When the receiver is lifted, the circuit closes and electricity flows. This current is -48V 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 won't 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 occu r; this refers to both ends of the circuit being seized at the same time, so that you pick up the phon e
and there is a caller on the line at the same moment, by coincidence.
Ground-Start Signaling Ground-start signaling is an adaptation of loop-start. Instead of the circuit being closed only at the phone end, both ends of the circuit have the capability to detect current, and both ends can request and confirm the use of the circuit. This is achieved by both end s being able to grou nd one of the wires in the circuit. These wires (or leads) are referred to as Tip and Ring. These terms date back to the use of 1/4-inch jacks with a positive contact at the tip and a negative conductor in the ring. The advantage is that it makes glare much less likely, and consequently ground-start is appropriate for trunk connections that are heavily used. However, it is very rare to see a ground-start trunk in a VoIP network or indeed in any new trunk deployment.
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FIGURE 9 Ground-Start Signaling
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. Th e 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.
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E&M Signaling Variously 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 signaling leads 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 the signaling-unit side; this is the CO, channel-bank, or Cisco gateway E&M interface. The E lead is used to indicate to the trunk side that the signaling-unit side has gone off-hook; conversely, the M lead is used to indicate to the signaling-unit side 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 can be 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 adva ntage of Delay Dial is that some e quip ment is not ready to receive digits instantly, even though it has sent the wink; the delay compensates for this.
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Introduction to Digital Circuits Digital circuits have the chief advantage of allowing a much higher density of calls on a given physical connection; an analog 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 30 calls. (For these values, we are assuming the calls are not compressed; more on this later). CCS circuits are designated as PRI Tl, PRI El, 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 to digital is performed by a codec. The following sections discuss the conversion of analog to digital.
Digitizing Analog Signals There 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 sample Sampling could be done any number of times per second; the more samples taken per second, the higher the audio quality, but the amount of digital data produced is much larger. Nyquist's theorem states that the sampling interval should be 2x the highest frequency of the sample to produce acceptable audio quality during playback. Because the highest frequency in human speech that we want to reproduce in telephony is around 4000 Hz, the sampling rate for standard to llquality digital voice is 8000 intervals per second. By contrast, CD music audio, which must encode both much higher and much lower frequencies, samples at about 192,000 times per second.
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Quantizing refers to making a digital approximation of an analog waveform. Imagine drawing an arc on a chessboard; if you had to define the arc using only the square it was in for each row (segment) and column (interval), you would end up with a stepped pattern that was sort of close to the original arc but not exact. This is exactly the process that happens wi th quantization: the codec chooses a segment value that is as close as possible to the analog value at the interval it was sampled, but it cannot be exact. To make the quantization more accurate, each sample is divided into 16 intervals that are adjusted to more closely match the sampled wave. Furthermore, the segments are actually more fine-grained at the origin than at the high and low ranges. T his is because m ost of the huma n speec h we are trying to cap ture accurately is in this center range of the scale; there are fewer sounds at the very highest and lowest values.
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FIGURE 11 Quantizing the Digital Sample
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.
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NOTE The determination of voice quality is based on the Mean Opinion Score (MOS). This is a subjec tive measurement, created by gathering the opinion of live human listeners. A sample recording is played, and the listeners give it a score out of 5, where 5 is best. The same sample is played using different compression or process ing methods and scored again. Because MOS is so subjective, other quality measurements exist that are more empirical and more accu rate. For reference, stan dard PCM encoding (G.711) scores 4.1, and G.729 scores 3.92.
Compression is not a required step, but it is often done to save bandwidth in VoIP environments. The two main types of compression we are concerned with are the following: •
Adaptive Differen Differential tial PCM (ADPC M): This method does not send entire code words, but instead sends a smaller
code that represents the difference between this word and the last one sent. This is not commonly used today, because it produces lower voice quality and compresses down only to about 16 kbps. •
Conjugate Structure Algebraic Code Excited Linear Prediction (CS_ACELP): As the name suggests, this is
more complex compression. Based on a dictionary or codebook of known sounds made by a standardized American male voice, the digital sample is analyzed and compared to the dictionary. The dictionary code that is the closest to the sample is sent. The codebook is constantly learning. The output of this compression is typically 8 kbps—with very little degradation of voice quality. This compression is widely used in VoIP.
Time Division Multiplexing (TDM) TDM is the primary technology used in traditional digital voice; it is also extensively used in data circuits. The basic premise is to take pieces of multiple streams of digital data and interleave them on a single transmission medium.
T1 Circuits On a Tl circuit, there are up to 24 channels available for voice. 64k from conversation 1 is loaded into the first Tl 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
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24 frames are grouped together (called Extended SuperFrame or ESF). Tls are typically full duplex, with two wires sending and the other two wires receiving.
E1 Circuit s An El is very similar to a T l. Th ere are 32 chan nels, of which 30 can be used for voice. (The other two are used for framing and signaling, respectively.) The 32 channels are grouped together as a frame, and 16 frames are grouped together as a multiframe. El circuits are common in Europe and Mexico, with some El services becoming available in the United States.
Channel Associated Signaling (CAS)—T1 Although the 64 k channels of a Tl are intended to carry digitized voice, we must also be able to transmit signaling in for mation, such as on-hook and off-hook, addressing, and so forth. In CAS circuits, the least significant bit of each channel in every sixth frame is "stolen" to generate signaling bit strings. SF implementation takes 12 frames and creates a SuperFrame. Using one bit per channel in every sixth frame gives two 12-bit signaling strings (known as A and B) per SuperFrame. 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 more advanced supervisory functions. Because CAS takes one bit from each channel in every sixth frame, it is known as Robbed Bit Signaling (RBS). Using RBS means that a slight degradation occurs in voice quality because every sixth frame has only 7 instead of 8 bits to represent the sample; however, this is not generally a perceptible degradation.
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Channel Associated Signaling (CAS)—T1 El signaling is slightly different. In an El CAS circuit, the first channel (channel 0 or timeslot 1) is reserved for f raming information. The 17th channel (channel 16 or timeslot 17) contains signaling information—no bits are robbed from the individual channels. Timeslots 2-16 and 18-32 carry the voice data. Each channel has specific bits in timeslot 17 fo r signaling. This means that although El CAS does not use RBS, it is still considered CAS; however, the signaling is ou tof-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 or BRI implementation. The full 64 k of bandwidth per channel is available for voice; instead of generating ABCD bits, a protocol known as Q.931 is used out-of-band in a separate channel for signaling. An ISDN PRI Tl provides 23 voice channels of 64 k each (called Bearer or B channels) and one 64 k D (for Data) channel (timeslot 24) for signaling. An ISDN PRI El provides 30 B channels and 1 D channel (timeslot 17); an ISDN BRI circuit provides two 64 k B channe ls and one D channel of 16 k.
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Understanding VoIP The elements of traditional telephony—status, address and supervisory signaling, digitization, and so on—must have functional parallels in the VoIP world for systems to function as people expect them to, and more importantly, for VoIP to interact 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 Packetization IP networks move data around in small pieces known as packets. Because we know how to digitize our voice, it now becomes just another binary payload to move around in a packet. VoIP uses Digital Signal Processors (DSP) for the codec functions. The digitized voice is then packaged in an appropriate protocol structure to move it through the IP infrastructure.
DSPs DSPs are specialized chips that perform high-speed codec functions. DSPs are found in the IP phones to encode the analog speech of the user and to decode the digitized contents of the packets arriving from the other end of the call. DSPs are also used on IOS gateways at the interface to PSTN circuits, to change from a digital circuit to packetized voice, or from an analog circuit to packetized voice. DSPs also change from one codec to another, allow conferencing and call park, and other telephony features. DSPs are a vital component of a VoIP system. Different chip types have varying capacities, but the general rule is that you want as many DSP resources available to you as possible. The DSP calculator on cisco.com 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, then by UDP, then by IP. A Layer 2 header of the correct format is applied; the type obviously depends on the link technology in use by each router interface. A single voice call generates two one-way RTP/UDP/IP packet streams. UDP provides multiplexing and checksum capability; RTP provides payload identification, timestamps, and sequence numbering. © 20 08 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
Payload identification allows us to treat voice traffic differently from video, for example, simply by looking for the RTP header label, simplifying our configuration tasks. Timestamping and sequence numbering allows VoIP devices to reorder RTP packets that arrived out of sequence and play them back with the same timing in which they were recorded, elimi nating delay s or jerkin ess. Th ere 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 quality of the RTP stream, a llowing devic es to record events such as pa cket 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). Because sampling is occurring at 8000 times per second, 20 msec gives us 160 samples. If we divide 8000 by 160, we see that w e are generating 50 packets with 160 bytes of payload, per second, for a one-way voice stream. If we use compressio n, we can squeeze the 160-byte payloa d down to 20 bytes using the G.729 codec. We still have 160 samples, still 20 msec of audio, but reduced payload size.
Codecs The 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
•
Anne x B variant: Allows the use of Voice Activity Detectio n and Comfort Noise Ge neratio n; can be applied to G.729 or G.729-A
The values for bandwidth shown do not include the Layer 3 and Layer 2 overhead; the actual bandwidth used by a sin gle (one-way) voice stream can be significantly larger. The following tables summarize the additional overhead added by packetization and Layer 2 encapsulation (assume 50 packets per second (pps):
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Bandwidth Calculation, Without Layer 2 Codec
G.711
G.729
Voice Payload
160 Bytes
20 Bytes
RTP Header
12 Bytes
12 Bytes
UDP Header
8 Bytes
8 Bytes
IP Header
20 Bytes
20 Bytes
Total Before Layer 2
200 Bytes
60 Bytes
Total Bitrate @ 50 pps
80,000 bps (80 kbps)
24,000 bps (24 kbps)
Bandwidth Calculation, With Layer 2 Layer 2 Type
G.71 1 = 20 0 Bytes/p acket
G.72 9 = 60 Bytes/p acket
Ethernet
18 Bytes
18 Bytes
Multilink PPP
6 Bytes
Frame Relay FRF. 12
6 Bytes 6 Bytes
6 Bytes
Total incl. Layer 2
218 Bytes
206 Bytes
206 Bytes
78 Bytes
66 Bytes
66 Bytes
Total Bitrate incl. Layer 2 (@ 50 pps)
87.200 (87.2 kbps)
82,400 (82.4 kbps)
82,400 (82.4 kbps)
31,200 (31.2 kbps)
26,400 (26.4 kbps)
26,400 (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 bandwid th just for heade r transmission on a slow link. The reco mme nde d solution is to use Comp resse d 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.)
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Bandwidth Calculation, Using cRTP Codec
G.711
G.729
Voice Payload
160 Bytes
20 Bytes
cRTP header w/ chksum
4 Bytes
4 Bytes
cRTP header no chksum Total before Layer 2:
2 Bytes 164 Bytes
162 Bytes
2 Bytes 24 Bytes
22 Bytes
Multilink PPP or Frame Relay FRF. 12
6 Bytes
Total WAN bandw idth @50 pps incl. Layer 2:
68000 bps (68 kbps)
6 Bytes 67,200 bps (67.2 kbps)
12,000 bps (12 kbps)
11,200 (11.2 kbps)
Voice Activity Detection (VAD) Phone conversations on average include about 35% 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 VAD feature can be enabled, causing the voice stream to be stopped during periods of silence. The theory here is that the ban d width otherwise used for silence can be reclaimed for voice or data transmission. VAD also adds Comfort Noise Generation (CNG), which fills in the dead silence created by the stopped voice flow with white noise. VAD should not be taken 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.
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Additional DS P Functions In addition to digitizing voice, DSP 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.
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Introducing VoIP Signaling Protocols VoIP signaling protocols are responsible for call setup, maintenance, and teardown. A number of different protocols are in use—some standards-based, others proprietary, and each with advantages and disadvantages. The following sections introduce the signaling protocols you should know about, including SCCP, H.323, MGCP, and SIP.
VoIP Signaling Protocols VoIP signaling protocols han dle the call setup, mainte nanc e, and teardow n functions of VoIP calls. It is importan t to keep in mind that the signaling functions are an entirely separate packet stream from the actual voice bearer path (RTP). The signaling protocol in use must pass the supervisory, informational, and address information expected in any telephony system. VoIP signaling protocols are either peer-to-peer or client-server; in the case of peer-to-peer protocols, the endpoints have the intelligence to perform the call-control signaling themselves, Client-server protocols send event notifications to the call agent (the Unified CM server) and receive instructions on what actions to perform in response. The following table summarizes the characteristics of the four signaling protocols dealt with here. Protocol
Standard?
Inter-Vendor Compatibi lity
Implemented on Gateway s
Imple mented on Cisco IP Phones
Operating Mode
H.323
Yes--I TU
Very Good
Yes
No
Peer-to-Peer
MGCP
Yes--IETF
Good
Yes
No
Client/Server
SIP
Yes--IETF
Basic
Yes
Yes; also third-party phone s
Peer-to-Peer
SCCP
No- -Cisco Proprietary
Cisco only
Some
Cisco IP Phones only
Client/Server
H.323 H.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 LAN
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environments, it is a long-established and stable protocol very suitable for intervendor compatibility. H.323 is supported by all Cisco voice gateways and CM platforms as well as some third-party video endpoints.
MGCP Media Gateway Control Protocol is a lightweight client/server protocol for PSTN gateways and some clients. It is simple to configure and allows the call agent to control the MGCP gateway, eliminating the need for expensive gateways with intelligence and complex configurations. The gateway reports events such as a trunk going off-hook, and the call agent instructs the gateway on what to do; the gateway has no local dial plan because all call routing decisions are made at the call agent and relayed to the MGCP gateway. MGCP is not as widely implemented as SIP or H.323. MGCP is not supported by Unified CM Express or the Smart Business Communication System.
SIP Session Initiation Protocol is an IETF standard that uses peer-to-peer signaling. It is very similar in structure and syntax to HTTP, and because it is text-based, it is relatively simple to debug and troubleshoot. SIP can use multiple transport layer protocols and can support security and proxy functions. SIP is an evolving standard that currently provides basic telephony functionality; further developments and extensions to the standard will soon make it feature-comparable with SCCP. One of its most important capabilities is creating SIP trunks to IP Telephony service providers, replacing or enhancing traditional TDM PSTN connections.
SCCP Skinny Client Control Protocol is a Cisco-proprietary signaling protocol used in a client-server manner between Unified CM 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 to events such as a phone going off-hook or a digit being dialed. SCCP is the default signaling protocol for all Cisco IP phones, although many also support SIP; SIP does not yet support the full feature set available to SCCP phones. All Cisco Unified Communications call agents (CM, CM Express, and the 500 Series) and some gateways support SCCP.
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Connecting a VoIP System to a Service Provider Network A VoIP system that can place calls only to other devices on the IP network is only marginally useful; we still need to place calls out to the PSTN, and to do so we need to connect to a phone service provider, whether via traditional TDM links or ITSP connections. The device that acts as the interface to the PSTN is the voice gateway; it provides the physic al connection and logical translation between two or more different network technologies.
Understanding Gateways, Voice Ports, and Dial Peers The following sec tions establish some te rms of reference .
Gateways In the Cisco Unified Communications architecture, a gateway is typically a voice-enabled router with an appropriate voice port installed and configured. Gateways can have both analog and digital voice port connections, including analog FXO, FXS, and E&M or digital Tl/El or PRI interfaces.
Call Legs A call leg is the inbound or outbound call path as it passes through the gateway. As the call comes into the gateway, it is associated with an inbound port. (This is the inbound call leg.) As the call is sent out another gateway port, this creates the outbound call leg. There will be inbound and outbound call legs at each gateway router.
Dial Peers A dial peer is a pointer to an endpoint, identified by an address (a pattern of digits). Cisco gateways support two types of dial peers: POTS and VoIP. POTS dial peers are addressed with PSTN phone numbers, and VoIP dial peers are addressed by IP addresses. Dial peers identify the source and destination endpoints of call legs; an inbound call leg is matched to a dial peer, and the outbound call leg is routed to a destination dial peer. Depending on the direction of the call, the dial peers may be POTS inbound and VoIP outbound, vice versa, or possibly both VoIP. It is unlikely but not impossible that the © 2 008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.
inbound and outbound dial peers would both be POTS. Each dial peer also defines attributes such as the codec to use, QoS settings, 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: Gateway(config)#dial-peer voice 10 pots Gateway(config-dialpeer)#destination-pattern Gateway(config-dialpeer)#port
8675309
1/0/1
The number assigned to dial peers is arbitrary, although dial peer 0 exists by default and cannot be deleted. The keyword pots 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 the destination pattern. Th e destination-pattern command associates a phone number with a dial peer. The specified pattern can be a specific phone number (as above, 8675309) or an expression that defines a range of numbers. The router uses the patterns to decide which dial peer (and associated physical port) it should route a call to. The following table briefly explains destination-pattern syntax. Character
Meaning
+
The preceding digit is repeated one or more times.
* and #
NOT wildcards; these are valid DTM F digits.
, (comma)
Inserts a one-secon d 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.
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Connecting a VoIP System to a Service Provider Network Configuring VoIP dial peers is equally simple. Examine the following configuration: Gateway(config)#dial-peer voice 20 voip G a t e w a y( c o n f i g - d i a l pe e r ) #
destina tion-pa ttern
Gateway(config-dialpeer)#
session
target
4. .. .
ipv4:10.1.1.2
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 The default dial peer 0 cannot be deleted or modified. It does not negotiate services such as VAD, DTMF Relay, or TCL applications. The dial peer 0 configuration for inbound VoIP calls contains the following: •
Any codec
•
VAD enabled
•
No RSVP Support
•
Fax-rate voice
1 . Look for the incoming called-number command in a dial peer that matches the called number or DNIS string of the inbound leg.
2. Look for the answer-address command in a dial peer that matches the calling number or ANI string of the inbound call leg.
3. Look for the destination-pattern command in a dial peer that matches the calling number or ANI string of the inbound call leg. 4. Look for the POTS dial peer port command that matches the voice port of the incoming call (POTS dial peers only). 5. If all of the above fail to match, match against Default Dial Peer 0 as a last resort. The default dial peer 0 config for inbound POTS calls includes the following: • no ivr application 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.
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Consider the following configuration: dial-peer voice 10 voip destination-pattern session
target
.T
ipv4:10.10.10 .1
i dial peer voice 20 voip destination-pattern session
target
867[2-3]...
ipv4:10.10.20 .1
! dial-peer voice 30 voip destination-pattern 8674... session targe t
ipv4:
1 0 . 1 0 . 3 0.1 0 .1
i dial-peer voice 40 voip destination-pattern session
target
8675309
ipv4:10.10.40.1
Given 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 numb er 867-4309 will match dial peer 30 (first (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 Providers As VoIP technology matured and stabilized, telephone service providers began extending VoIP connectivity to their customers, allowing for simple, flexible connection alternatives to traditional TDM links. Internet Telephony Service © 20 08 Cisco Systems Inc. All rights reserved. This publication publication is protected by copyright. Please see page 147 for more details.
Providers (ITSP) connections are typically much less expensive, available in smaller bandwidth increments than Tl or PRI links, and can route nonvoice data traffic concurrently. QoS configuration is supported (and in fact is required for proper 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 to the provider, who then routes calls to their PSTN connection, usually with a toll-minimizing route that dramatically reduces long-distance costs to the customer.
Understanding Call Setup and Digit Manipulation Successfully completing a phone call requires that the correct digits are sent to the terminating device, whether on the VoIP network or the PSTN. PSTN calls are typically more complex because of the varying local and international requirements for the number of digits required to route the call. Over and above this basic requirement are the additional complex ities impose d by requiremen ts of the business: we ma y want to change our ANI number, a dd or strip strip access codes, compensate for undesirable default behavior, or build specialized functionality for our particular purposes. This section deals with digit manipulation and troubleshooting.
Digit Consumption and Forwarding Some strange things happen when an IOS gateway matches a dial peer for an outbound call leg and forwards the dialed digits to the terminating device. For POTS dial peers, the gateway consumes (meaning strips away) the left-justified digits that exactly match the dial-peer destination pattern and forwards only the wildcard-matched digits to the terminating device. Clearly, this could cause problems if the PSTN were to receive only 4 digits, as in this example: dial-peer
voice
20
destination-pattern port
pots 867....
1/0:1
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With this configuration, if the dialed number was 867-5309, the gateway would forward only 5309 (the wildcard matches), 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 Collection The 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 overlapping digits, this can cause calls to be misrouted, as in the following example: Dial- peer voice 1 voip Destination Session
pattern
target
555
ipv4:10.1.1 .1
! Dial-peer voice 2 voip Destination-pattern Session
target
5552112
ipv4:10.2.2.2
If the user dials 555-2112, dial peer 1 will exactly match at the third digit, the call will be immediately routed using dial peer 1, and only the collected digits of 555 will be forwarded. We solve the problem by changing the configuration as shown next: Dial- peer voice 1 voip Destination Session
pattern
target
555....
ipv4:10.1.1 .1
! Dial-peer voice 2 voip Destination-pattern Session
target
5552112
ipv4:10.2.2.2
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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. Dial peer 1 is also a match, but beca use of the wildcards, the destinatio n pattern matche s 10,00 0 possible numbers (000 0 through 9999); it is not as close a match as dial peer 2.
Digit Manipulation Sometimes we need to add, change, or remove digits before the call is placed. We do this to avoid inconveniencing users or to match the dialed digit requirements of a gateway or the PSTN. We have several methods of modifying the digit string, as described in the following sections.
prefix Th e prefix dial-peer command adds digits to the beginning of the string after the outbound dial peer is matched but before passing digits to the destination. An example of its use is a POTS dial peer with 2... as the destination pattern. If the user dials 2112, the default behavior is for the POTS dial peer to forward only 112. Adding the command prefix 6 0 4 5 5 5 2 forces the router to prepend the additional digits required to route the call over the PSTN: dial-peer
voice
20
destination-pattern prefix port
pots 2...
6045552 1/0/0
forward-digits forward-digits: This dial-peer command forces the specified number of digits to be forwarded, whether the digits were
exact match or wildcard matches, overriding the default behavior of stripping the exact matches. You can specify a number of digits to forward (as shown in the example that follows) or use forward-digits all to force all dialed digits to be forwarded. dial-peer
voice
20
destination-pattern
pots 5552...
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forward-digits 7 port
1/0/0
Number
Expansion
n u m - e x p : The number expansion table is a global command that either expands an extension (perhaps a 4-digit extension
into a full 10-digit PSTN number) or completely replaces one number with another. This command is applied before the outbound dial peer is matched, so there must be a configured dial peer that matches the expanded number for the call to be forwarded. n u m- ex p 2 . . . dial-peer
5552 ...
voice
20
pots
destination-pattern port
5552...
1/0/0
Translation
Rules
voice translation-rule: This global command configures number translation profiles to allow us to alter the ANI, DNIS,
or redirect number for a call. Using the command is a three-step process:
1. Define the translation rule globally: voice tran sla tion -ru le rul e 1
1
/555/ /867/
Th e rule command defines a pattern to match (in this case 555) and a pattern to change the matched digits to (in this case 867). The match and replace patterns are identified and separated by the "/" characters that begin and end the patterns.
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2. Create the voice translation profile containing the translate instruction (the options are [calledlcallingl redirect-calledlredirect-target], and reference the rule we just defined by number. In this example we are translat ing the called number: v o i c e t r a n s l a t i o n - p r o f i l e JENNY translate
called
1
3. Apply the profile to one or more dial peers, either inbound or outbound: dial-peer
voice
desc ript ion
20
pots
t r a n s l a t e d t o J e nn y
translation-profile outgoing JENNY destination-pattern port
5552...
1/0/0
Translation rules use regular expression syntax, which can be quite complex. The following table defines the characters used, and examples follow.
Cisco Regular Expression Characters for Voice Translation Rules Character
Description
Matches any single character. \ (mat ch)
In the match phrase: Escape the special meaning of the next character.
\(rep lace )
In the repl ace phra se: Refe renc e a set num ber from the matc h phra se. Match the expression at the beginning of the digit string.
A
$ /
Match the expression at the end of the digit string. Identifies the start and end of both the match and replace phrases.
[0-9]
Match a single character in a list.
[ 0-9]
Do not match a single character specified in the list.
A
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Cisco Regular Expression Characters for Voice Translation Rules *
Repeat the previous expression 0 or more times.
+
Repeat the previous expression 1 or more times.
?
Repeat the previous expression 0 or 1 time.
()
Identifies a set in the match expression.
continued
Example 1: rul e 1
/123/
/456/
The first set of forward slashes defines the match phrase; the second set defines the replace phrase. This expression means "match 123 and replace it with 456." Thus: •
123 is replaced with 456
•
6123 is replaced with 6456
•
1234 is replaced with 4564
•
1234123 is replaced with 4564123 (only the first instance of the match is replaced)
Example 2: voice rule
translation?rule 1
1
/ 4 0 . . . / / 6 66 66 60 60 B 0/ 0/ A
This example replaces any five-digit number that begins with "40" with the number "6666000". Example 3: voice
translation?rule
/ \ ( 8 6 7 \ ) \ ( . . . . \ ) / A
1
/ 5 5 5 \ 2 /
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This example means: "If the number starts with 867 and is followed by any four other digits, change the 867 to 555 and replace the other four digits with the digits in Set 2 of the match." Remember that the forward slashes define the match and replace phrases; the backslashes mean "the next character is not part of what to match"; the round brackets indicate which sets of characters in the matched number to keep in the replaced number. The sets are numbered starting with 1, so the first set of round brackets is 1, and the second is 2 (as in this example).
Private Line Automatic Ringdown (PLAR) PLAR creates a permanent association between a voice port and a destination number (or voice port). When PLAR is configured, going off-hook on that voice port automatically dials the pattern specified by the connection plar command. The caller does not hear a dial tone and does not have to dial a number. Think of PLAR as a hotline; pick up the Batline and you get Batman without having to dial. The following shows a simple PLAR configuration that will call 867-5309 when the phone goes off-hook: voice
port
connection
1/0/0 plar
8675309
Troubleshooting Dial Plans and Dial Peers The following sections discuss some of the commands available to troubleshoot your configuration.
show dial-peer voice To display information for voice dial peers, use the show dial-peer voice command in user EXEC or privileged EXEC mode. show dial-peer voice
[number | summary]
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Syntax Description number
(Optional) A specific voice dial peer. Output displays detailed information about that dial peer.
summary
(Optional) Output displays a short summary of each voice dial peer.
If both the name argument and s u m m a r y keyword are omitted, output displays detailed information about all voice dial peers. The following is sample output from this command for a VoIP dial peer: Router#
show
dial-peer
voice
101
VoiceOverIpPeer101 peer type = voice, information type = voice, description
=
tag = 6001,
'',
destin atio n-pat tern
a n s w e rr- a d d r e s s =
'' ,
=
"6001' ,
preference=0,
CLID Restriction = None CLID Network Number = "
1
CLID Second Number s e n t CLID Override RDNIS = disabled, s ou ou r ce ce c a r r i e r - i d = s ou ou r ce ce t r u n k - g r o u p - l a b e l
target car ri er- id = =
'',
target trunk-gr oup-label
= '' ,
numbering Type = "unknown' group = 6001, Admin state is up, incoming ca ll ed - num number ber =
Operation state is up,
connections /m connections /m ax im um = 0 / u n l i m i t e d ,
DTMF Relay = disabled,