IPTX | IP Telephony Express Volume 1 Version 4.0 Student Guide Text Part Number: 97-2439-01Table of Contents Volume 1 Course Introduction 1 Overview 1 Learner Skills and Knowledge 1 Course Goal and Objectives 2 Course Flow 3 Additional References 4 Cisco Glossary of Terms 4 Cisco Unified CallManager Express Fundamentals 1-1 Overview 14 Module Objectives 141 Describing Key Features of Cisco Unified CallManager Express and Cisco Unity Express 4:3 Overview 1-3 Objectives 13 What Is Cisco Unified CallManager Express? 1-4 What Is Cisco Unity Express? 1-6 How Do Cisco Unified CallManager Express and Cisco Unity Express Work? 1-9 Licensing 114 Summary 1-19 Explaining Differences Between Traditional Telephony and VoIP 1-21 Overview 1-21 Objectives 1-21 Traditional Telephony 1-22 CO Switching Systems 1-25 PCM Theory 1-29 Basic Voice Encoding: Converting Digital to Analog 1-30 The Nyquist Theorem 1-31 Quantization 1-32 Coder-Decoders and Compression 1-34 Example 1.37 Example 1-37 Encapsulating Voice in IP Packets 1-39 Example 1-39 Example 1-40 IP Voice Header Compression 1-41 RTP Packet Components 1-42 Summary 1-44 Understanding VoIP Challenges and Solutions 1-45 Overview 1-45 Objectives 1-45 Requirements of Voice in an IP Internetwork 1-46 Example 1-46 Example 1-48 Example 1-51 Challenges in VoIP 1-52 Bandwidth Requirements in VoIP 1-54 Example 1-55 Example 1-57 Example 1-59 1-60 SummaryDescribing the Cisco Unified CallManager Express Voice Packet Handling Methods 1-61 Overview 1-61 Objectives 1-61 IP Phone Calls 1-62 Packet Forwarding, Voice Packet Priority, and RTP Stream Information 168 WAN Call Setup 1-70 Summary 4-74 Module Summary 1-75 References 1-76 Module Self-Check 177 Module Self-Check Answer Key 1-81 Cisco Unified CallManager Express Configuration 2-1 Overview 24 Module Objectives 24 Understanding Cisco Unified CallManager Express Features and Functionality 2-3 Overview 23 Objectives 23 Key Benefits and Features 24 Phone Features 25 System Features 26 Trunk Features 27 Voice-Mail Features 27 Supported Platforms and Telephones 28 Example 29 ‘Supported Protocols and Integration Options 2-31 Cisco Unified CallManager Express Requirements 2:38 Cisco Unified CallManager Express Restrictions 2:39 Summary 2-41 Configuring Cisco Unified CallManager Express Network Parameters 2-43 Overview 2-43 Objectives 2.43 Voice VLANs 2.44 Configuring Voice VLANs 2-46 DHCP Service Setup 2-50 DHCP Relay Server 257 Network Time Protocol 2-60 Transcoding 2-65 Summary 2-84 Understanding the IP Phone Registration Process 2-85 Overview 2-85 Objectives 2-85 IP Phone Firmware and XML Configuration Files 2-86 IP Phone Information 2-92 Downloading and Registration 2-93 Summary 2-99 Defining Ephone-dn and Ephone 2-104 Overview 2-101 Objectives 2-101 What Are an Ephone and an Ephone-dn? 2-102 Ephone-dn 2-103 Ephone 2-109 ‘Types of Ephone-dns 2-115 Summary 2-131 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Introducing SIP and Cisco Unified CallManager Express 2-133 Overview 2-133 Objectives 2-133 Cisco Unified CallManager Express and SIP 2-134 SIP IP Phone Prerequisites 2-135 SIP IP Phone System Settings 2-139 Voice Register Directory Number 2-143 Voice Register Pool 2145 Loading SIP Firmware on a SCCP IP Phone 2-449 Loading SCCP Firmware on a SIP IP Phone 2-153 Summary 2-154 Describing Cisco Unified CallManager Express Files 2-155 Overview 2-155 Objectives 2-155 Cisco Unified CallManager Express Files 2-156 Bundled Cisco Unified CallManager Express Files 2-457 Individual Cisco Unified CallManager Express Files 2-159 GUI Files 2-160 Cisco Unified CallManager Express-TAPI Integration 2-162 Additional Files 2-163 Summary 2-164 Understanding Initial Phone Setup_ 2-165 Overview 2-165 Objectives 2-165 Setting Up Phones in a Cisco Unified CallManager Express System 2-166 Manual Phone Setup 2-167 Example 2-470 Example 2-472 Example 24174 Example: Manual Setup of Cisco Unified CallManager Express 2477 Partially Automated Phone Setup 2178 Automated Phone Setup 2-181 Quick Configuration Too! 2-185 Optional Parameters 2-490 Rebooting Cisco Unified CallManager Express Phones 2-494 Setup Troubleshooting Tips 2-497 Verifying Cisco Unified CallManager Express Phone Configuration 2-202 Summary 2-203 Module Summary 2-205 References 2-206 Module Self-Check 2.207 Module Self-Check Answer Key 2-213 PSTN Interface and Voice Dial Peer Configuration 3-1 Overview 34 Module Objectives 34 Understanding Analog and Digital Voice Interfaces 3:3 Overview 33 Objectives 33 Local-Loop Connections 34 Analog Voice Interfaces 35 Channel Associated Signaling Systems: T1 3-8 Channel Associated Signaling Systems: £1 3-10 Common-Channel Signaling Systems 342 PRI and BRI 3-43 ‘Summary 3-44 © 2006 Cisco Systems, Inc. IP Telephony Express (IPTX) v4.0 iiConfiguring Analog and Digital Voice Interfaces A5 Overview 3-15 Objectives 3-16 Foreign Exchange Station Port Configuration 347 Configuration Parameters 3-18 Example 3-19 Foreign Exchange Office Port Configuration 3-20 Configuration Parameters 3-20 Ear and Mouth Port Configuration 3-22 Configuration Parameters 3-22 Timers and Timing 3-24 Configuration Parameters 324 Digital Voice Port Configuration 3-26 Configuration Parameters 3-26 Channel Associated Signaling Configuration 3-29 Common-Channel Signaling: BRI 3.31 Common-Channel Signaling: PRI 3:37 Summary 3-43 Configuring Dial Peers 3-45 Overview 3.45 Objectives 3-45 What Is a Dial Peer? 3-46 Plain Old Telephone Service Dial Peers 3-49 Example 3-50 VoIP Dial Peers 351 Example 3-52 Example 3-52 Destination Pattern Options 3.53 Example 3-55 Matching Inbound Dial Peers 3-56 What Is the Default Dial Peer? 3-58 Example 3-59 Matching Outbound Dial Peers 3-60 Example 3-60 Summary 3-61 Understanding Call Setup and Digit Manipulation 3-63 Overview 3-63 Objectives 3-63 Digit Collection and Consumption 3-64 Example 3-65 What Is Digit Manipulation? 3-67 Example 3-69 Private Line, Automatic Ringdown 371 ‘Summary 3-73 Module Summary 3-75 References 375 Module Self-Check 3-76 Module Self-Check Answer Key 3-79 wv IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.IPTX | Course Introduction Overview IP Telephony Express (PTX) v4.0 provides an understanding of Cisco Unified CallManager Express and Cisco Unity Express and of the challenges that you face when configuring and deploying the systems. The course presents Cisco Systems solutions and implementation considerations for addressing those challenges. Learner Skills and Knowledge This subtopic lists the skills and knowledge that learners must possess to benefit fully from thi course. Learner Skills and Knowledge Working knowledge of LANs Working knowledge WANs Working knowledge of IP switching and routing Basic internetworking skills Knowledge of traditional PSTN operations and technologiesCourse Goal and Objectives This topic describes the course goal and objectives. oal “To configure Cisco Unified CallManager Express and Cisco Unity Express and apply VoIP technologies to a Cisco Unified CallManager Express and Cisco Unity Express environment” IP Telephony Express Upon completing this course, you will be able to meet these objectives = Describe the key features and functionality of the Cisco Unified CallManager Express = Explain the differences between traditional voice and VoIP = Configure Cisco Unified CallManager Express to support IP phones = Configure analog voice interfaces, digital voice interfaces, and dial peers to set up VolP communications = Configure the GUI features, phone features, TAPI, and network management = Install and configure Cisco Unity Express = Plan Cisco Unified CallManager Express and Cisco Unity Express deployment 2 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IrCourse Flow This topic presents the suggested flow of the course mater Course Flow Diagram ce Day2 PEE Pees Pe Course Introduction PSTN interface | Configuration of | Cisco Unity Cisco Unified ‘and Voice Dist | Additional Cisco | Express Auto | CallManager Peer Configuration] Unified Attendantand | Express and Cisco Unified CallManager | _ Voico-Mail ‘Cisco Unity CallManager Express Features | Configuration Express: Express Network Design Fundamentals Configuration of PSTN Interface and] Additional Cisco Voice Dial Peer Unified Cisco Unity Gisco Unified | Configuration | CallManager | Express Auto Calthanager Express Features | Attendant and xpress | |“Configuration of | Cisco Unity Voice-Mail Configuration | additional Cisco | Express Auto| Configuration Unified Attendant and CallManager ‘Voice Mail Express Features | Configuration The schedule reflects the recommended structure for this course. This structure allows enough time for the instructor to present the course information and for you to work through the lab activities. The exact timing of the subject materials and labs depends on the pace of your specific class. ‘© 2006 Cisco Systems, Inc. Course IntroductionAdditional References This topic presents the Cisco icons and symbols that are used in this course, as well as information on where to find additional technical references. Cisco Icons and Symbols ~p~ GBB 4 am Switch Generic Softswitch uv “$ Voice-Enabled Communications Server ers Web Browser 2 Cisco Glossary of Terms ig ~S S Ss wy POX Firewall {Right and Lett) PC. Muttayer Switen, with Text, without Text, Laptop and Subdued Cisco Unified CallManager Express Workgroup Voice-Enabled ATM Switeh For additional information on Cisco terminology, refer to the Cisco Internetworking Terms and Acronyms glossary of terms at http://www cisco.com/univered/ce/td/doe/cisintwk/ita/index. htm. 4 IP Telephony Express (IPTX) v4.0. © 2006 Cisco Systems, Inc.Module 1 | Cisco Unified CallManager Express Fundamentals Overview Cisco Unified CallManager Express is an integrated call processing solution that is based on Cisco midrange access routers using Cisco IOS software. Cisco Unified CallManager Expre delivers telephony services for up to 240 users ina small and medium-sized offices. It is part of the Cisco Unified Communications solution and works in conjunction with the extended Cisco Systems product portfolio, including routers, data switches, public switched telephone network (PSTN) gateways, gatekeepers, Cisco Unity voice mail, and analog terminal adapters. Cisco Unified CallManager Express delivers a robust set of telephony features that are similar to those commonly used by business users. Cisco Unified CallManager Express is an optional feature of Cisco IOS sofiware and is available on a wide range of Cisco access routers that support as many as 240 IP phones. This solution allows customers to take advantage of the benefits of IP communication without the higher costs and complexity of deploying a server- based solution. Furthermore, because the solution is based on the Cisco access router and Cisco 10S software, it is simple to deploy and manage, especially for customers who already use Cisco IOS software products. Cisco Unity Express offers local voice-mail and automated attendant capabilities for IP phone users in a small office or branch location that is connected to Cisco Unified CallManager or Cisco Unified CallManager Express. Cisco Unity Express is fully integrated into the branch office router, either on a Cisco Unity Express Network Module or on a Cisco Unity Express ‘Advanced Integration Module (AIM). Module Objectives Upon completing this module, you will be able to describe the features, positioning strategies, and deployment models of Cisco Unified CallManager Express and contrast traditional telephony and VoIP. This ability includes being able to meet these objectives: Describe the key features and functionality of the Cisco Unified CallManager Express system m= Describe the differences between traditional telephony technology and the equivalent mechanisms in VoIP= Describe the challenges and solutions associated with VoIP delivery in LANs and WANs = Describe the Cisco Unified CallManager Express voice packet handling methods. 4-2 IP Telephony Express (IPTX) v4.0 (© 2006 Cisco Systems, IncLesson 1 | Describing Key Features of Cisco Unified CallManager Express and Cisco Unity Express Overview This lesson describes the key features and functionality of Cisco Unified CallManager Express and Cisco Unity Express. This information includes the licensing scheme and the effect of licensing on activation of features. You are directed to the Cisco website for up-to-date information on licensing. Objectives Upon completing this lesson, you will be able to explain the differences between traditional voice and VoIP. This ability includes being able to meet these objectives @ Define Cisco Unified CallManager Express = Define Cisco Unity Express = Describe the functionality of Cisco Unified CallManager Express and Cisco Unity Express . Describe licensing requirements and the effect of licensing on feature activationWhat Is Cisco Unified CallManager Express? This topic describes the Cisco Unified CallManager Express system. What Is Cisco Unified CallManager Lelie Cisco Unified CallManager Express + Call processing for small and medium-sized deployments + VoIP integrated solution + Up to 240 IP phones + Cisco IOS software-based solution Cisco Unified CallManager Express is an integrated call processing solution that is based on Cisco midrange access routers that use Cisco IOS software to deliver telephony services for 10 to 240 users in small and medium-sized offices. Cisco Unified CallManager Express is part of the Cisco Unified Communications solution and works in conjunction with the extended Cisco Systems product portfolio, including routers, data switches, public switched telephone network (PSTN) gateways, gatekeepers. Cisco Unity voice mail, and analog telephone adapters (ATAS). Cisco Unified CallManager Express delivers a robust set of telephony features that are similar to those commonly used by businesses. Cisco Unified CallManager Express is an optional feature of Cisco 1OS software and is available on a wide range of Cisco access routers that support as many as 240 IP phones. This option allows customers to take advantage of the benefits of IP communications without the higher cost and complexity of deploying a server- based solution, Because the solution is based on the Cisco access router and Cisco IOS. software, it is simple to deploy and manage, especially for customers who already use Cisco IOS software products. Cisco Unified CallManager Express allows customers to scale IP telephony to a small or branch office site with a solution that is easy to deploy, administer, and maintain. 4-4 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, inc.What Is Cisco Unified CallManager Express? (Cont.) + Integrated Services Routers + Multiservice Platform Routers. Cisco Unified CallManager Express enables the large portfolio of Cisco Multiservice Access Routers and Integrated Services Routers to deliver features that are similar to low-end PBX and key system features, creating a cost-effective, highly reliable, feature-rich IP communications solution for the small office. Cisco Unified CallManager Express supports a new generation of intelligent IP phones with robust display capabilities. End users can easily customize these phones based on their changing needs. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 15What Is Cisco Unity Express? This topic describes the Cisco Unity Express system, What Is Cisco Unity Express? Voice mail and Auto Attendant for small offices and branch offices Fully integrated into the Cisco 3800 and 2800 Series Integrated Services Routers and the Cisco 2600 Series, 2691, and 3700 Series Multiservice Platforms Three form factors: Cisco Unity Express Network Module, Cisco Unity Express Network Module Enhanced Capacity, and Cisco Unity Express Advanced Integration Module Two call-control options: Cisco Unified CallManager Express and Cisco Unified CallManager Cisco Unity Express offers local voice mail and automated attendant capabilities for IP phone users connected to Cisco Unified CallManager or Cisco Unified CallManager Express in a small office or branch location. You can fully integrate Cisco Unity Express into the branch office router on a Cisco Unity Express Network Module, a Cisco Unity Express Network Module Enhanced Capacity, or a Cisco Unity Express Advanced Integration Module (AIM). 16 IP Telephony Express (IPTX) v4.0. © 2006 Cisco Systems, IncWhat Is Cisco Unity Express? (Cont.) Cisco Unity Express Network Module or Cisco Unity Express Network Module Enhanced Capacity Voice message storage: 300 hours with Cisco Unity Express Network Module Enhanced Capacity or 100 hours with Cisco Unity Express Network Module + Hard drive storage Available as of Reloase 1.0 Cisco Unity Express AIM . Cisco 3800, 3700, aE ‘2800, 2600xM, and 2691 routers + Voice message storage: 8 hours with 512-MB flash card or 14 hours with #-G8 flash card + 512.MB or 1-68 compact fash storage + Industia-qualty ash with prolonged life and wear leveling + Available a8 of Release 1.1 Cisco Unity Express is currently available in Cisco Unity Express Network Module, Cisco Unity Express Network Module Enhanced Capacity, and Cisco Unity Express AIM. The network-based modules are the more scalable and powerful modules, but they do require the whole slot in the chassis in which they reside, The Cisco Unity Express AIM resides on the motherboard of the router; it conserves valuable network module slots and expands the number of Cisco router platforms that can support both voice-mail and analog interfaces, lowering the cost of an entry-level system. ‘The storage is either a hard drive in the Cisco Unity Express Network Module or Cisco Unity Express Network Module Enhanced Capacity or a flash card in the Cisco Unity Express AIM. ‘The hard drive in the Cisco Unity Express Network Module and Cisco Unity Express Network Module Enhanced Capacity is not a field-replaceable unit (FRU). The whole module must be sent back to Cisco if'a hard drive failure occurs. Flash memory has a limited lifetime and must be replaced after a certain number of writes has occurred. In a typical environment, replacement will be needed every three to five years, Note The flash module is an industrial-grade flash; off-the-shelf flash cannot be used, © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-7What Is Cisco Unity Express? (Cont.) Voice mail: + Supports up to 250 subscriber mailboxes on the Cisco Unity Express Network Module Enhanced Capacity + Supports up to 100 subscriber mailboxes on the Cisco Unity Express Network Module + Supports up to 50 subscriber mailboxes on the Cisco Unity Express AIM + Storage is configurable per subscriber Auto Attendant: + Has up to five active Auto Attendants per system + Offers fully customizable script-driven menu structure and menu nesting + Has time-of-day and day-of-week call treatment + Business hours can be defined + Holidays can be defined Voice mail is essential in most enterprises. Voice mail enables messages to be left for subscribers when they are busy or do not answer a call in a specified amount of time An automated attendant is a device that automatically answers calls with an interactive recording and allows callers to route their call to the desired person or department by entering the appropriate extension using the telephone keypad. Businesses can customize the greeting by adding information such as hours and directions, Cisco Unity Express supports a built-in automated attendant along with its voice-mail capabilities. IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.How Do Cisco Unified CallManager Express and Cisco Unity Express Work? This topic describes how Cisco Unified CallManager Express and Cisco Unity Express work. How Do Cisco Unified CallManager Express and Cisco Unity Express Work? Cisco Unified CallManager Express is a Cisco IOS software-based call control agent. Register Register Phones register with Cisco Unified CallManager Express and are then under its control. = The Cisco Unified CallManager Express system provides PBX-like features and functions for IP phones. These features stem from the concept of a centralized point of control and intelligence. The Cisco Unified CallManager Express router provides all of the call control and intelligence needed for IP phones to place and receive calls. In a Cisco Unified CallManager Express deployment, the IP phones are not capable of setting up a call by themselves. In fact, the administrator completely controls the IP phones by using the Cisco Unified CallManager Express system and instructs the phones on how to place and receive calls The IP phones boot and register with Cisco Unified CallManager Express. If you properly configure Cisco Unified CallManager Express, calls can be set up and torn down to and from the IP phones. The IP phones and the Cisco Unified CallManager Express router use Skinny Client Control Protocol (SCCP) to communicate. Note Registration across a WAN is not supported. The IP phones must be on the local LAN with the Cisco Unified CallManager Express router. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals, 14How Do Cisco Unified CallManager Express and Cisco Unity Express Work? (Cont.) Cisco Unified CallManager Express is a Cisco IOS software-based call control agent. Phone A places call to phone B. —_—— Phone A RTP Phone B Call control is centralized on Cisco Unified CallManager Express. When a call is placed between two IP phones that are under the control of Cisco Unified CallManager Express, SCCP is used to set up the call. SCCP does not go between the two IP phones, only between the IP phone and the Cisco Unified CallManager Express system. After the call is set up, Real-Time Transport Protocol (RTP) is used to carry the audio stream. RTP is a. common protocol that is used to carry time-sensitive traffic, such as voice and real-time video. RTP is carried inside a User Datagram Protocol (UDP) segment, which is then carried inside an IP packet This is the sequence of events for a phone call Step1 Phone A picks up the handset and dials the number of phone B. Step2 The dialed digits are sent through SCCP to Cisco Unified CallManager Express. Step 3 Cisco Unified CallManager Express knows the location of phone B (because of the registration) and its status (busy, on hook, off hook), Step4 Assuming that phone B is on hook (available), Cisco Unified CallManager Express sends an SCCP message to tell phone B about the incoming call and to tell it to ring, Step5 Phone B is answered. Step6 Cisco Unified CallManager Express informs each IP phone about the settings of the other phone and instructs both phones to construct RTP connections. Step7 The IP phones construct two one-way RTP connections for the voice to travel across, one for the voice of phone A to travel to B, and one for the voice of phone B to travel to A. Step8 The call takes place. Step9 Phone B hangs up and an SCCP message is sent to Cisco Unified CallManager Express. “10 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Ino.Step 10 Cisco Unified CallManager Express sends an SCCP message to phone A telling it that the call has been disconnected. How Do Cisco Unified CallManager Express and Cisco Unity Express Work? (Cont.) Connection(s) to PSTN + Analog igital Cisco Unified CallManager Express can act as a PSTN gateway as well as manage the IP phones. There are various types of connections to the PSTN, including digital and analog. Thi type of connection depends on the density of connections that is needed, the technology that i available in the region, the cost of the connections, and the interfaces on the router. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals,How Do Cisco Unified CallManager Express and Cisco Unity Express Work? (Cont.) ‘Step 4: Cisco Unified CallManager Express uses SIP to set up a call to the Unity Express module. igco Unity Cisco Unified CallManager Express Ca + ‘Step 5: The call is set up, and voice flows between Cisco Unity Express and the PSTN gateway function of the router. ae Stop 2: An SCOP message causes 2 the IP phone to Z ey & a No ring, Step 1: A call arrives from the newer PSTN that maps through DID to Secure within he the IP phone whose extension Set time value. is 1000. Cisco Unified CallManager Express and Cisco Unity Express interact when Cisco Unified CallManager Express determines that a call needs to go either to voice mail or to the Auto Attendant. The figure shows a call from the PSTN being forwarded to voice mail using the following steps: Step 1 Step 2 Step 3 Step 4 Step 5 A call arrives from the PSTN and, based on the called number, is mapped using direct inward dialing (DID) to the internal extension 1000. Cisco Unified CallManager Express sends an SCCP message to the IP phone and causes the IP phone to ring. The timeout value for no answer to a forwarded call is exceeded, so Cisco Unified CallManager Express follows the forwarding instructions and forwards the call to the Cisco Unity Express voice-mail pilot number, A session initiation protocol (SIP) message is sent to the IP address of the Cisco Unity Express module to set up a voice connection using one of the virtual voice ports. The Cisco Unity Express module has a free virtual voice port and answers the call via an SIP message that goes back to Cisco Unified CallManager Express. Two unidirectional RTP streams are created between the PSTN gateway function of the router and Cisco Unity Express. 12 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IncHow Do Cisco Unified CallManager Express and Cisco Unity Express Work? (Cont.) Cisco Uni CallManager Express Cluster SIP PSTN Gateway and IP-to-IP Gateway Functionality If one Cisco Unified CallManager Express system needs to set up a call to an IP phone that is under the control of another Cisco Unified CallManager Express system, then the H.323 protocol needs to be used between the Cisco Unified CallManager Express systems. This configuration allows for many different deployments of Cisco Unified CallManager Express to be integrated through an IP-based WAN link. The PSTN gateway function can be performed on the Cisco Unified CallManager Expres router or on a separate stand-alone gateway. If'a separate PSTN gateway is used, the add functionality of an IP-to-IP gateway can also be run on the router. This configuration would enable the ability to translate between H.323 and SIP. nal Note A local PSTN is needed for each site for, at the very least, 9-1-1 emergency calls. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-13,Licensing This topic describes the licensing for Cisco Unified CallManager Express and Cisco Unity Express. Licensing Licensing for Cisco Unified CallManager Express: + Capable Cisco IOS image + Feature license for number of phones + Seat license per phone up to 240 Licensing for Cisco Unity Express: + License for 12 mailboxes is included. + Additional licenses can be purchased for up to 250 mailboxes total on the Cisco Unity Express Network Module Enhanced Capacity. Additional licenses can be purchased for up to 100 mailboxes total on the Cisco Unity Express Network Module. + Additional licenses can be purchased for up to 50 mailboxes total on the Cisco Unity Express Advanced Integration Module. Both Cisco Unified CallManager Express and Cisco Unity Express have licensing requirements. For Cisco Unified CallManager Express, first a capable Cisco IOS image must be installed on the router, and then the proper feature license must be purchased. The feature license defines how many phones will be controlled with the Cisco Unified CallManager Express software. The various feature licenses are as follows: =) FL-CCME-SMALL (up to 24 users) = ~FL-CCME-36 (up to 36 users) = FL-CCME-MEDIUM (up to 48 users) = FL-CCME-72 (up to 72 users) m= FL-CCME-96 (up to 96 users) = = FL-CCME-120 (up to 120 users) . . . . FL-CCME-144 (up to 144 users) FL-CCME-168 (up to 168 users) FL-CCME-192 (up to 192 users) FL-CCME-240 (up to 240 users) In addition to the feature license, each analog phone controlled by an ATA and each IP phone requires a seat license, The Cisco Unified CallManager Express scat license is fully transferable to a Cisco Unified CallManager seat license. There are 12 licensed user mailboxes included with the Cisco Unity Express module when it is ordered, If more than 12 mailboxes are needed or desired, a new license file must be installed on the Cisco Unity Express module. 114 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.ES) S eT are tty Offer savings and ease of ordering when compared with ordering each of the components separately Flexible base package with option to add additional service modules to provide customer with complete solution + Include Cisco IOS SP Services for voice gateway services and features Can be easily upgraded Include DSP modules to support PSTN-to-IP connectivity Allow country-specific PSTN analog or digit to meet customer needs Include Cisco IP Communications features license Offer flexibility to choose appropriate Cisco Unity Express module for voice mail ean Cisco offers a broad choice of IP communications solutions for growing businesses. For businesses with a need for secure IP data routing with full-service voice capabilities, Cisco Unified CallManager Express Bundles offer an affordable entry point into Cisco Unified Communications, These turnkey communications solutions support up to 240 phones and deliver feature-rich call processing with integrated routing and switching, as well as optional voice mail and Auto Attendant. ‘Small businesses can expect to realize the following returns on their Cisco Unified CallManager Express Bundles investment: = Cost savings and productivity enhancements: Cisco Unified CallManager Express Bundles are an affordable entry point into a converged IP environment that delivers cost savings and productivity enhancements. m= Investment protection: Cisco Unified CallManager Express Bundles are cost-effective, and they integrate with legacy voice investments while allowing you to migrate to a Cisco Unified Communications system. = Ease of management: The bundle components are integrated within a single chassis, resulting in turnkey installation and streamlined system management with a common GUL = Growth: Designed to respond to your dynamic business needs, Cisco Unified CallManager Express Bundles can be easily upgraded to support advanced voice applications and additional users, The complete portfolio of the Cisco Unified Communications solution scales to support up to 30,000 devices. = Support: With an excellent record of accomplishment in supporting mission-critical voice applications, Cisco and its certified partners provide full life-cycle support to deliver Cisco Unified CallManager Express Bundles for a maximum return on investment. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-15,Cisco offers a range of bundles tailored to meet the needs of your business. Each bundle includes a Cisco IP access router for secure data routing, Cisco Unified CallManager Express software to support IP telephony, Cisco IOS SP Services software for voice gateway services, digital signal processor (DSP) chips for PSTN calls, and memory. Cisco Unity Express may be added to the bundle for voice mail and Auto Attendant capabilities. The base Cisco Unified CallManager Express Bundles are designed to meet the diverse needs of businesses worldwide It is necessary to add the country-specific digital or analog trunk interfaces that are required to connect to the PSTN or host PBX. To complete the solution, add Cisco IP phones and Cisco Catalyst data switches that support inline power. The various bundles include the following SKUs: = 2801-CCME/K9: 2801-V router, DSP resources for 8 calls, 24 Cisco Unified CallManager Express seats, and Cisco IOS SP Services @ = 2811-CCME/K9: 2811-V router, DSP resources for 16 calls, 36 Cisco Unified CallManager Express seats, and Cisco IOS SP Services m= 2821-CCME/K9: 2821-V router, DSP resources for 32 calls, 48 Cisco Unified CallManager Express seats, and Cisco IOS SP Services = 2851-CCME/K9: 2851-V router, DSP resources for 48 calls, 96 Cisco Unified CallManager Express seats, and Cisco IOS SP Services @ = 3825-CCME/K9: 3825-V/K9 router, DSP resources for 64 calls, 168 Cisco Unified CallManager Express seats, and Cisco IOS SP Services ™ 3845-CCME/K9: 3845-V/K9 router, DSP resources for 64 calls, 240 Cisco Unified CallManager Express seats, and Cisco IOS SP Services 16 IP Telephony Express (|PTX) v4.0 © 2006 Cisco Systems, Inc.Release Compatibility ary en Le ce Personal 2 25 50 100 250 ‘Genorat-dolivery 3 10 5 20 5 mailboxes isco Unity Express 100 700 Network Module: Hours of storage, Cisco Unity Express Network Module Enhanced Capacity: Hours of storage, Cisco Unity Express Network Module: Number of ports Cisco Unity Express Network Module Enhanced Capacity: Number of ports There are four Cisco Unity Express license levels available on the Cisco Unity Express Network Module and Cisco Unity Express Network Module Enhanced Capacity. The hardware associated with Cisco Unity Express, the Cisco Unity Express Network Module and the Cisco Unity Express AIM, must be purchased with an accompanying license. Hardware and software are packaged together. Mailbox licenses are purchased separately, with the exception of the 12- mailbox license level that is included in the price of the hardware-software bundle. Therefore, a minimum of 12 mailboxes must be ordered with each Cisco Unity Express purchase. Cisco Unity Express license files, such as Cisco IOS software, can be downloaded from http://cisco.com and installed on any number of systems for which a license was purchased without change to the file itself, When a license is purchased, when software from Cisco is used, or both, a contractual obligation is created. The subscriber must abide by the terms spelled out in the license agreement, including prohibitions regarding unauthorized replication of the software and modification to the mailbox level of the license. The capacity limitations on ports, subscribers, and mailboxes depend on whether Cisco Unity Express is running on a network module or an advanced integration module and is controlled by the license that is installed on the Cisco Unity Express application. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals. 1-11,Release Compatibility (Cont.) 2 ry 50) 100 250 Mailboxes Mailboxes Mailboxes Mailboxes Mailboxes Cisco Unity Express AIM, Not 11GB: Hours of storage supported Cisco Unity Expross AIM, Not 1 GB, 2600 Series and ‘supported 2691: Number of ports Cisco Unity Expross AIM, Not 1.GB, 2800, 3700 and supported 3800: Number of ports "Not recommended because of port blocking and mailbox size li ‘There are three Cisco Unity Express license levels available with the Cisco Unity Express AIM in the 1-GB model. The module comes with a 12-mailbox license by default and may be upgraded to 25 or $0 mailboxes. You can license the Cisco Unity Express Network Module at 12, 25, 50, or 100 mailboxes, and you can license the Cisco Unity Express Network Module Enhanced Capacity at 12, 25, 50,100, 150, 200, and 250 mailboxes. When the Cisco Unity Express AIM is located in the chassis of a Cisco 2600 Series or 2691 Multiservice Platform Router, itis limited to a maximum of four simultaneous ports at any one time. This configuration presents some port blocking issues that may be manifested when the number of mailboxes approaches the upper limit of 50. 118 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc.Summary This topic summarizes the key points that were discussed in this lesson. Eira + Cisco Unified CallManager Express is an optional feature of Cisco IOS software and is available on a wide range of Cisco access routers that support as many as 240 phones. + Cisco Unified CallManager Express provides call processing for IP phones using SCCP. + Cisco Unity Express provides voice mail and Auto Attendant for the small office or branch office. + Cisco Unity Express is fully integrated into Cisco 2600 Series and 2691 Multiservice Platform routers, 2600 and 3800 Series Integrated Services Routers, and 3700 Series Multiservice Access Routers. (© 2006 Cisco Systems, Ine Cisco Unified Callanager Express Fundamentals 1-19,Lesson 2| Explaining Differences Between Traditional Telephony and VoIP Overview This lesson explains the differences between traditional voice and VoIP. This discussion includes traditional telephony, pulse code modulation (PCM) theory, and the basics of voice digitization. It also includes a discussion of the various compression schemes that are used to transport voice using less bandwidth, using coder-decoder (codec) attributes, and encapsulating voice in IP packets. In addition, the use of compressed Real-Time Transport Protocol (cRTP) headers, including when and when not to use them, is discussed Objectives Upon completing this lesson, you will be able to explain the differences between traditional voice and VoIP. This ability includes being able to meet these objectives: m= Identify the components, processes, and features of traditional telephony networks that provide end-to-end call functionality m= Identify the steps for converting analog signals to digital signals and the steps for converting digital signals to analog signals; state the purpose of the Nyquist theorem; and explain quantization = Explain voice compression and codec standards; name two types of voice compression techniques; and list three common voice compression standards and their bandwidth requirements = Describe the functions of RTP and RTCP as they relate to a VoIP network = Describe how IP voice headers are compressed using ¢RTP and how header size is reduced to efficiently carry voice across the network using VoIP protocols and cRTPTraditional Telephony This topic introduces the components of traditional telephony networks. It describes how central office (CO) switches function and how they make switching decisions. It explores PBX and key telephone system functionality in environments today. The topic also discusses the I signaling types: supervisory, address, and informational. Basic Components of a Telephony Network ‘A number of components m st be in place for an end-to-end call to succeed. These components are shown in the figure and include the following: Edge devices Local loops Private or CO switches Trunks, Edge Devices There are two types of edge devices that a telephony network uses: Analog telephones: Analog telephones are most common in home, small busines: small office, home office (SOHO) environments. A direct connection to the public switched telephone network (PSTN) is usually made by using analog telephones. Proprietary analog telephones are occasionally used in conjunction with a PBX. These phones provide additional functions, such as speakerphone, volume control, PBX message- waiting indicator, call on hold, and personalized ringing. and Digital telephones: Digital telephones contain hardware to convert analog voice into a digitized stream. Larger corporate environments with PBXs generally use digital telephones. Digital telephones are typically proprietary; that is, they work with the PBX or key system of that vendor only. 1-22 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Local Loops A local loop is a physically wired, electrical interface between the telephone company network and each telephone that it services. Typically, a single wire pair carries a single conversation. A home or small business may have multiple local loops. Private or CO Switches The CO switch terminates the local loop and handles signaling, digit collection, call routing, call setup, and call teardown. A PBX switch is a privately owned switch located at the customer site. A PBX can interface with other components to provide additional services, such as voice mail. Trunks The primary function of a trunk is to provide the path between two switches. Several common, trunk types include these: = Tie trunk: A dedicated circuit that connects PBXs directly = CO trunk: A direct connection between a local CO and a PBX = Interoffice trunk: A circuit that connects two local telephone company COs ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-28Central Office Switches The figure shows a typical CO switch environment. The CO switch terminates the local loop and makes the initial call routing decision. The call routing function will forward a call to one of the following destinations: = Another end-user telephone if it is connected to the same CO = Another CO switch = A tandem switch The CO switch enables the telephone to work with the following components: = Battery: The battery is the source of power to both the circuit and the telephone—it determines the status of the circuit. When the handset is lifted to let current flow, the telephone company provides the source that powers the circuit and the telephone, Because the telephone company powers the telephone from the CO, your electric company power is separate from your phone company power. Power outages at your electric company will not affect the basic telephone unless the phone company relies on the electricity provider for power. Cordless telephones require a supplementary power source that the subscriber supplies. Cordless telephones lose function during a power outage unless there is a local backup power supply. = Current detector: The current detector monitors the status of a circuit by detecting whether it is open or closed. See the “Current Flow in a Typical Telephone” table, 4-24 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc.Current Flow in a Typical Telephone Handset Circuit Current Flow SE EE ee On cradle (On hook or open circuit No Off cradie Off hook or closed circuit Yes = Dial tone generator: When the digit register is ready, the dial tone generator produces a dial tone to acknowledge the request for service. = Digit register: The digit register receives the dialed digits. = Ring generator: When the switch detects a call for a specific subscriber, the ring generator alerts the called party by sending a ring signal to that subscriber. ‘You must configure a PBX connection to a CO switch that matches the signaling of the CO switch. This configuration ensures that the switch and the PBX can detect on-hook and off hook states and dialed digits coming from either direction. CO Switching Systems Switching systems provide three primary functions = Call setup, routing, and teardown = Call supervision = Customer IDs and telephone numbers A switch at the CO will switch calls between locally terminated telephones. If a call recipient is not locally connected, the CO switch decides where to send the call based on its call routing table. The call then travels over a trunk to another CO or to an intermediate switch that may belong to an interexchange carrier (IXC). Although intermediate switches do not provide a dial tone, they act as hubs to connect the other switches and provide interswitch call routing PSTN calls are traditionally circuit-switched. This design guarantees the end-to-end path and resources. When the PSTN sends a call from one switch to another, the same resource is associated with the call until the call terminates. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-25What Is a PBX? Line Cara ‘Trunk Card Local ‘A PBX is a smaller, privately owned version of a CO switch that the telephone company uses. In a corporate environment, where large numbers of staff need access to each other and to the outside, individual telephone lines are not economically viable. Most businesses have a PBX telephone system, key telephone system, or Centrex service. Offices with more than 50 telephones or handsets will likely choose a PBX type of solution to connect users, both in- house and to the PSTN PBXs come in a variety of sizes, typically from 20 to 20,000 stations. Selecting a PBX device is important because it has an average life span of 7 to 10 years. All PBXs offer a basic set of calling features. Optional software provides additional capabilities. The figure illustrates the internal components of a PBX. It connects to telephone handsets using line cards and to the local exchange using trunk cards. A PBX has three major components: = Terminal interface: The terminal interface provides the connection between terminals and PBX features that reside in the control complex. Terminals can include telephone handsets, trunks, and lines. Common PBX features include dial tone and ringing = Switching network: The switching network provides the transmission path between two or more terminals in a conversation, such as when two telephones within an office ‘communicate over the switching network, = Control complex: The control complex provides the logic, memory, and processing for call setup, call supervision, and call disconnection. 1-26 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IncWhat Is a Key System? Local Exchange Termination Blocks Trunks. ‘Connector Block Main Distibution Frame ‘Small organizations and branch offices often use a key system because a PBX device has more functionality than they require. For example, unlike the central answering position that is required for a PBX, a key system enables small businesses to have distributed answering from any telephone. Today, key systems are either analog or digital and are microprocessor-based. Key systems are usually installed in offices with 30 to 40 users and can be scaled to support more than 100 users. A key system has three major components: = Key Service Unit (KSU): Holds the system switching components, power, intercom, line and station cards, and system logic m= System software: Provides the operating system and calling-feature software Telephones (instruments or handsets): Allow the user to choose a free line and dial out, usually by pressing a button on the telephone © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-2Basic Call Setup Pat Step 2 rs Called Number = 555.0123, ia esa Local Trunk, ee ree Soe Loop rue) Network Signaling Pe er! Call signaling, in its most basic form, is the capacity of a user to communicate a need for service to a network. The call signaling process requires the ability to detect a request for termination of service, to send addressing information, and to provide progress reports to the initiating party. This functionality corresponds to the three call signaling types: supervisory, address, and informational The figure shows the three major steps in an end-to-end call: Step1 Local signaling—originating side: The user signals the switch by going off hook and sending dialed digits through the local loop. Step2 Network signaling: The switch makes a routing decision and signals the next, or terminating, switch using setup messages sent across a trunk. Step 3 Local signaling—terminating side: The terminating switch signals the call recipient by sending ringing voltage through the local loop to the recipient telephone. 1-28 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc.PCM Theory This topic describes the process of converting analog signals to digital signals and converting digital signals back to analog signals. The topic also describes the Nyquist theorem, which is the basis for digital signal technology, and explains quantization and its techniques. Digitizing Analog Signals Sample the analog signal regularly. Quantize the sample. Encode the value into a binary expression. (Optional) Compress the samples to reduce bandwidth (multiplexing). Digitizing speech was a project first undertaken by the Bell System in the 1950s. The original purpose of digitizing speech was to deploy more voice circuits with a smaller number of wires. This technology evolved into the T| and EI transmission methods of today. The result was pulse code modulation (PCM). PCM is the process in which a signal is sampled, and the magnitude of each sample is quantized and digitized for transmission (usually in 8-bit form). In conventional PCM, the analog data may be compressed before being digitized. However, after it is digitized, the PCM signal cannot be subjected to digital compression before being multiplexed into the data stream. To convert an analog signal to a digital signal, the steps shown in the table must be performed. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-2Analog-to-Digital Signal Conversion Step | Procedure Description 1. ‘Sample the analog signal regularly. The sampling rate must be two times the highest frequency to produce playback that appears neither choppy nor too smooth. Quantize the sample. Quantization consists of a scale made up of eight major divisions, or chords. Each chord is subdivided into 16 equally spaced steps. The chords are not equally spaced, but are actually finest near the origin. Steps are equal within the chords, but different when they are compared between the chords. Finer graduations at the origin result in less distortion for low-level tones. Encode the value into 8-bit digital form. PBX output is a continuous analog voice waveform. 11 digital voice is a snapshot of the wave, encoded in ones and zeros. 4. Compress the samples to reduce bandwidth Although not essential to the conversion of analog signals to digital, signal compression is widely used. to reduce bandwidth Note Step 4 is optional. There are three components in the analog-to-digital conversion process: Sampling: Sample the analog signal at periodic intervals. The output of sampling is a pulse n: Match the PAM signal to a segmented scale. This scale measures the amplitude (height) of the PAM signal and assigns an integer value to define that amplitude . amplitude modulation (PAM) signal = Qua . Encoding: Convert the integer base-10 number to a binary number. The output of encoding is a binary expression in which each bit is either a | (pulse) or a 0 (no pulse). This three-step process is repeated 8000 times per second for telephone voice channel service. Use the optional fourth step—compression—to save bandwidth. This optional step allows a single channel to carry more voice calls. Note The most commonly used method for converting analog to digital is PCM, Basic Voice Encoding: Converting Digital to Analog After the receiving terminal at the far end receives the digital PCM signal, it must convert the PCM signal back into an analog signal. The process of converting digital signals back into analog signals consists of two parts, decoding and filtering: = Decoding: The received 8-bit word is decoded to recover the number that defines the amplitude of that sample. This information is used to rebuild a PAM signal of the original amplitude. This process is simply the reverse of the analog-to-digital conversion. = Filtering: The PAM signal passes through a properly designed filter, which reconstructs the original analog waveform from its digitally coded counterpart. 1-30 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Nyquist Theorem ‘Analog Audio Source ‘Sampling Stage The Nyquist Theorem Example Digital signal technology is based on the premise stated in the Nyquist theorem: When a signal is instantaneously sampled at the transmitter in regular intervals and has a rate of at least twice the highest channel frequency, then the samples will contain sufficient information to allow an accurate reconstruction of the signal at the receiver. ‘The human ear can sense sounds from 20 to 20,000 Hz, and speech encompasses sounds from about 200 to 9000 Hz. The telephone channel was designed to operate at about 300 to 3400 Hz. This economical range carries enough fidelity to allow callers to identify the party at the far end and sense mood. Nyquist decided to extend the digitization to 4000 Hz to capture any higher- frequency sounds that the telephone channel might deliver. Therefore, the highest frequency for voice is 4000 Hz, or 8000 samples per second. That frequency converts to one sample every 125 microseconds. If every sample is encoded in 8 bits, this calculation works out to be 8000 samples a second times 8 bits per sample. The result is that a digital voice conversation requires 64,000 bits per second. This fact explains why the original digital data circuits that carried digital voice, known as DS0s, are sized at 64,000 bits per second ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-31Quantization Voltage ‘Segment 2 q ‘Segment 1 Segment 0 Note: Each line represents 1/8000 of second. Quantization involves dividing the range of amplitude values in an analog signal sample into a set of discrete steps that are closest in value to the original analog signal. Each step is assigned a unique digital code word. The figure depicts quantization. In this example, the x-axis is time and the y- value (the PAM). xis is the voltage The voltage range is divided into 16 segments (0 to 7 positive and 0 to 7 negative). Starting with segment 0, each segment has fewer steps than the previous segment, which reduces the noise-to-signal ratio and makes it uniform. This segmentation also corresponds closely to the logarithmic behavior of the human ear. Ifa noise-to-signal ratio problem exists, it is resolved by using a logarithmic scale to convert PAM to PCM. 1-32 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IncQuantization Techniques + Linear — Uniform quantization + Logarithmic quantization ~ Compands the signal ~ Provides a more uniform signal-to-noise ratio + Two methods a-law (most countries) ~ mu-law (Canada, United States, and Japan) Linear sampling of analog signals causes small-amplitude signals to have a higher noise-to- signal ratio, which results in a poorer quality signal than a larger-amplitude signal. The Bell system developed the mu-law method of quantization, which is widely used in North America. The ITU modified the original mu-law method and created a-law, which is used in countries outside North America Mu-law and a-law provide a method of reducing the noise-to-signal ratio method. This method allows smaller step functions at lower amplitudes than at higher amplitudes. Both mu-law and a-law “compand” the signal; that is, they both compress the signal for transmission and then expand the signal back to its original form at the other end. Using mu-law or a-law results in a more accurate value for smaller amplitudes and uniform signal-to-noise quantization ratio across the input range. Both mu-law and a-law are linear approximations of a logarithmic input-output relationship. Both generate 64-kbps bit streams using 8-bit code words to segment and quantize levels within segments. There may be some difference between the original analog signal and the assigned quantization level. This is called “quantization error,” which is the source of distortion in digital transmission systems. Quantization error is any random disturbance or signal that interferes with the quality of the transmission or the signal itself. Note For communication between a mu-law country and an a-law country, the mu-law country must change its signaling to accommodate the a-law country. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-33,Coder-Decoders and Compression This topic describes two types of speech-coding schemes, waveform and source coding, and compares G.729 and G.729A compression. Voice-Compression Techniques + Waveform algorithms - PCM ~ ADPCM + Source algorithms. ~ LDCELP - CS-ACELP There are two voice-compression techniques. = Waveform algorithms (coders) function as follows: — Coders take sample analog signals at the rate of 8000 times per second. — Coders use predictive differential methods to reduce bandwidth, and this reduction strongly affects voice quality. — Coders do not take advantage of speech characteristics. = Source algorithms function as follows: — Voice coders (vocoders) convert analog speech into digital speech, using a specific compression scheme that is optimized for coding human speech: — _ Vocoders take advantage of speech characteristics. — Codebooks store specific predictive waveshapes of human speech. They match the speech, encode the phrases, decode the waveshapes at the receiver by looking up the coded phrase, and match the coded phrase to the stored waveshape in the receiver codebook. 1-34 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Example: Waveform Compression + PCM — Waveform coding scheme + ADPCM ~ Waveform coding scheme ~ Adaptive: automatic companding ~ Differential: changes encoded between samples only + ITU standards: ~ G.711 rate: 64 kbps = (2 x 4 kHz) x 8 bits/sample 2 x 4 kHz) x 4 bits/sample 2 x 4 kHz) x 3 bits/sample 2 x 4 kHz) x 2 bits/sample Standard PCM is known as ITU standard G.711 An adaptive differential PCM (ADPCM) coder is a waveform coder that encodes analog voice signals into digital signals to predict future encodings by looking at the immediate past. The adaptive feature of ADCPM reduces the number of bits per second that the PCM method requires to encode voice signals. ADPCM does so by taking 8000 samples per second of the analog voice signal and turning them into a linear PCM sample, ADPCM then calculates the predicted value of the next sample, based on the immediate past sample, and encodes the difference. The ADPCM process generates 4-bit words, therefore generating 16 specific bit patterns. The ADPCM algorithm from the ITU-T (formerly the CCITT) transmits all 16 possible bit patterns. The ADPCM algorithm from the ANSI uses 15 of the 16 possible bit patterns excluding the 0000 pattern, ‘The ITU standards for compression are as follows: = G,711 rate: 64 kbps = (2 * 4 kHz) * 8 bits per sample = G.726 rate: 32 kbps = (2 * 4 kHz) * 4 bits per sample = G.726 rate: = G.726 rate: 16 kbps 4 kbps 2* 4 kHz) * 3 bits per sample 2 4 kHz) * 2 bits per sample © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals. 1-35Example: Source Compression + CELP — Hybrid coding scheme + High-quality voice at low bit rates; processor-intensive + G.728: LDCELP—16 kbps + G.729: CS-ACELP—8 kbps ~ G.729A variant—8 kbps, less processor-intensive, allows more voice channels encoded per digital signal processor ~ Annex B variant—VAD and CNG Code excited linear prediction (CELP) compression transforms analog voice signals as follows: = The input to the coder is converted from an 8-bit PCM sample to a 16-bit linear PCM sample. = A codebook uses feedback to continuously learn and predict the voice waveform. = A white noise generator excites the coder. = The mathematical result (recipe) is sent to the far-end decoder for synthesis and generation of the voice waveform. Conjugate structure algebraic code excited linear prediction (CS-ACELP) is a variation of CELP that performs these functions: = Codes on 80-byte frames, which take approximately 10 ms to buffer and proces m= Adds a look-ahead of 5 ms. A look-ahead is a coding mechanism that continuously analyzes, learns, and predicts the next waveshape. ™ Adds noise reduction and pitch-synthesis filtering to processing requirements, Low-delay CELP (LDCELP) is similar to CS-ACELP except for these characteristics: = LDCELP uses a smaller codebook and operates at 16 kbps to minimize look-ahead delay, keeping it to 2 to 5 ms, m= The 10-bit codeword is produced from every five speech samples from the 8-kKHz input. = Four of these 10-bit codewords are called a “subframe,” which takes approximately 2.5 ms to encode. Two of these subframes are combined into a 5-ms block for transmission. 1-36 IP Telephony Express (IPTX) v4.0 (© 2006 Cisco Systems, IncExample The Annex B variant adds voice activity detection (VAD) in strict compliance with G.729B standards. When this codec variant is used, VAD is not tunable for music threshold. However, when Cisco VAD is configured, music threshold is tunable. G.729 and G.729A Comparison + Both are ITU standards. + Both are 8-kbps CS-ACELP. + 6.729 is more complex and processor-intensive. + 6.729 is slightly higher quality than G.729A. + Compression delay is the same (10 to 20 ms). + Annex B variant can be applied to either. G.729, G.729 Annex A (G.729A), G.729 Annex B (G.729B), and G.729A Annex B (G.729AB) are variations of CS-ACELP. There is little difference between the ITU recommendations for G.729 and G.729A. All of the platforms that support G.729 also support G.729A. G.729 is the compression algorithm that Cisco uses for high-quality 8-kbps voice. When G.729 is properly implemented, it sounds as good as the 32-kbps ADPCM. G.729 is a high- complexity, processor-intensive compression algorithm that monopolizes processing resources. Although G.729A is also an 8-kbps compression, it is not as processor-intensive as G.729. It is a medium-complexity variant of G.729 with slightly lower voice quality. The quality of G.729A is not as high as G.729 and is more susceptible to network irregularities such as delay, variation, and “tandeming.” Tandeming causes distortion that occurs when speech is coded, decoded, then coded and decoded again, much like the distortion that occurs when a videotape is repeatedly copied. Example On Cisco IOS gateways, you must use the variant (G.729 or G.729A) that is related to the codec complexity configuration on the voice card. This variant does not show up explicitly in the Cisco IOS command-line interface (CLI) codec choice. For example, the CLI does not display “g72918” (alpha code) as a codec option. However, if the voice card is defined as medium complexity, then the g729r8 option is the G.729A codec. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-37G.729B is a high-complexity algorithm, and G.729AB is a medium-complexity variant of G.729B with slightly lower voice quality. The difference between the G.729 and G.729B, codecs is that the G.729B codec provides built-in Internet Engineering Task Force (IETF) VAD and comfort noise generation (CNG). The following G.729 codec combinations interoperate: = G.729 and G.729A G.729 and G.729 G.729A and G.729A G.729B and G.729AB G.729B and G.729B G.729AB and G.729AB 1-38 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Encapsulating Voice in IP Packets This topic describes the functions of Real-Time Transport Protocol (RTP) and Real-Time Transport Control Protocol (RTCP) as they relate to the VoIP network. Real-Time Transport Protocol end network functions and delivery services for delay-sensitive, real-time data, such as voice and video + Works with queuing to prioritize voice traffic over other traffic + Services include: ~ Payload type identification ~ Sequence numbering ~ Time-stamping ~ Delivery monitoring RTP provides end-to-end network transport functions intended for applications that are transmitting real-time data, such as audio and video. The functions include payload type identification, sequence numbering, time-stamping, and delivery monitoring. RTP typically runs on top of User Datagram Protocol (UDP) to utilize the multiplexing and checksum services of that protocol. Although RTP is often used for unicast sessions, it is primarily designed for multicast sessions. In addition to defining the roles of sender and receiver, RTP also defines the roles of translator and mixer to support the multicast requirements. Example RTP is a critical component of VoIP because it enables the destination device to reorder and retime the voice packets before they are played out to the user. An RTP header contains a time stamp and a sequence number, which allow the receiving device to buffer and remove jitter and latency by synchronizing the packets to play back a continuous stream of sound. RTP uses sequence numbers to order the packets only. RTP does not request retransmission if a packet is lost. For more information on RTP, refer to RFC 3550. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-39,Real-Time Transport Control Protocol Monitors the quality of the data distribution and provides control information Provides feedback on current network conditions Allows hosts that are involved in an RTP session to exchange information about monitoring and controlling the session Provides a separate flow from RTP for UDP transport use RE RTCP monitors the quality of the data distribution and provides control information. RTCP provides the following feedback on current network conditions: = RTCP provides a mechanism for hosts involved in an RTP session to exchange information about monitoring and controlling the session. RTCP monitors the quality of such elements as packet count, packet loss, delay, and inter-arrival jitter. RTCP transmits packets as a percentage of session bandwidth, but at a specific rate of at least every 5 seconds. = The RTP standard states that the Network Time Protocol (NTP) time stamp is based on synchronized clocks. The corresponding RTP time stamp is randomly generated and based on data-packet sampling. Both NTP and RTP are included in RTCP packets by the sender of the data. = RTCP provides a separate flow from RTP for transport use by UDP. When a voice stream is assigned UDP port numbers, RTP is typically assigned an even-numbered port while RTCP is assigned the next odd-numbered port. Each voice call has four ports assigned: RTP plus RTCP in the transmit direction and RTP plus RTCP in the receive direction. Example Throughout the duration of each RTP call, the RTCP report packets are generated at least every 5 seconds. In the event of poor network conditions, a call may be disconnected because of high packet loss. When using a packet analyzer to view packets, a network administrator can check information in the RTCP header that includes packet count, octet count, number of packets lost, and jitter. The RTCP header information helps in determining why calls are disconnected. 1-40 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.IP Voice Header Compression The topic describes how IP voice headers are compressed using ¢RTP, and it describes when to use cRTP. RTP Header Compression Before RTP Header Compression ‘After RTP Header Compression 20 8 12 2010120 Zorg 2010120 Bytes Bytes Bytes Bytes ca Cag re RTP headers are Header almost always 2 bytes 40 Bytes tong when compressed. + RTP header compression saves bandwidth by compressing packet headers across WAN links. Given the number of multiple protocols that are necessary to transport voice over an IP network, the packet header can be large. You can use cRTP headers on a link-by-link bi save bandwidth. Using cRTP compresses the IP/UDP/RTP header from 40 bytes to 2 bytes without UDP checksums and from 40 bytes to 4 bytes with UDP checksums. RTP header compression is especially beneficial when the RTP payload size is small, such as with compressed audio payloads that are 20 bytes and 50 bytes. In addition, CRTP assumes that most of the fields in the IP/UDP/RTP header do not change or that the change is predictable. Static fields include source and destination IP addresses, source and destination UDP port numbers, and many other fields in all three headers. The table illustrates the cRTP process for those fields in which the change is predictable. RTP Stage What Happens The change is predictable. The sending side tracks the predicted change. “The predicted change is tracked. The sending side sends a hash of the header. “The receiving side predicts what the | The receiving side substitutes the original stored header and constant change is. calculates the changed fields. ‘An unexpected change occurs. | The sending side sends the entire header without compression. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-41RTP Packet Components When speech samples are framed every 20 ms in a packet voice environment that is using G.729, a payload of 20 bytes is generated. Without cRTP, the total packet size includes the following components: m IP header (20 bytes) = UDP header (8 bytes) = RTP header (12 bytes) m= Payload (20 bytes) The header is twice the size of the payload: IP/UDP/RTP (20 + 8 + 12 = 40 bytes) versus the payload (20 bytes), When packets are generated every 20 ms on a slow link, the header consumes a large portion of bandwidth. ‘As shown in the previous figure, RTP header compression reduces the header to 2 bytes. Now, instead of the header being twice the size of the payload, the payload is 10 times the size of the compressed header. 4-42 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Example When to Use RTP Header Compression + Congested WAN links + Slow links (less than 2 Mbps) + Bandwidth on a WAN interface that needs to be conserved = Congested WAN links m= Slow links (less than 2 Mbps) = Bandwidth on a WAN interface that needs to be conserved Compression works on a link-by-link basis and must be enabled for each link that has any of those conditions. You must enable compression on both sides of the link for proper results. Enabling compression on both ends of a low-bandwidth serial link can greatly reduce the network overhead if there is a significant volume of RTP traffic on that slow link. Note Compression adds to processing overhead. You must check resource availability on each device prior to turning on RTP header compression, If you want the router to compress RTP packets, use the ip rtp header-compression command, The ip rtp header-compression command defaults to active mode when it is configured. However, this command provides a passive mode setting in instances when you want the router to compress RTP packets only if it has received compressed RTP on that interface. For a Frame Relay interface, use the frame-relay ip rtp header-compression command. By default, the software supports 16 RTP header compression connections on an interface. Depending on the traffic on the interface, you can change the number of header compression connections with the ip rtp compression-connections number command. Note Do not use cRTP if the link is faster than 2 Mbps. © 2006 Cisco ‘Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-43Summary This topic summarizes the key points that were discussed in this lesson, Birla + Traditional telephony networks are composed of edge devices such as tolephones, local loops, switches, and trunks. CO switches terminate local loops and provide battery, current detection, dial tone, ring generation, and the digit registers. + The three parts of the analog-to-digital conversion process are sampling, quantization, and encoding. The two parts of the digital- to-analog conversion process are decoding and filtering, ‘Two techniques used for voice compression are waveform ‘compression and source compression, and the three common voice-compression standards are PCM, ADPCM, and CELP. + RTP carries packetized audio traffic over an IP network, and RTCP provides feedback on the quality of the call, including statistics on packet loss, dolay, and jitter. + RTP header compression compresses the IP/UDP/RTP header in an. RTP data packet from 40 bytes to approximately 2 to 4 bytes, most of the time. 1-44 IP Telephony Express (PTX) v4.0 1© 2006 Cisco Systems, ncLesson 3| Understanding VoIP Challenges and Solutions Overview This lesson discusses the challenges and solutions that are associated with VoIP delivery in LANs and WANs. This discussion includes requirements for voice delivery in an IP network, the challenges of VoIP, bandwidth requirements, and the need for quality of service (QoS). To understand the QoS issues that you will encounter, you need to be able to calculate the amount of bandwidth that will be consumed. Several variables that affect total bandwidth are explained as is the method for calculating and reducing total bandwidth. Objectives Upon completing this lesson, you will be able to discuss the challenges and solutions associatec with VoIP. This ability includes being able to meet these objectives: = Determine the best method for improving delivery of voice packets with minimal loss, delay, and jitter, taking into account the challenges associated with implementing VoIP solutions ® Discuss the challenges associated with voice delivery in an IP network m= List the bandwidth requirements for various codecs and data links and describe methods to reduce bandwidth consumptionRequirements of Voice in an IP Internetwork This topic lists problems associated with implementation of real-time voice traffic in a best- effort IP internetwork and discusses the causes of packet loss, end-to-end delay, and jitter delay in an IP internetwork. This topic also describes the methods that you can use to ensure consistent delivery and throughput of voice packets in an IP internetwork. Finally, this topic describes how Real-Time Transport Protocol (RTP) ensures consistent delivery order of voice packets in an IP internetwork | — + IP is connectionless. + IP provides multiple paths from source to destination. The traditional telephony network was originally designed to carry voice. The design of circuit- switched calls provides a guaranteed path and delay threshold between source and destination, The IP network was originally designed to carry data. Data networks were not designed to carry voice traffic. Data traffic is best-effort traffic and can withstand some amount of delay, jitter, and loss, while voice traffic is real-time traffic that requires a specified level of QoS. In the absence of any special QoS parameters, a voice packet is treated as just another data packet. The user must have a well-engineered network, end to end, when running delay-sensitive applications such as VoIP. Fine-tuning the network to adequately support VoIP involves a series of protocols and features that have to do with QoS. Example In the IP network shown in the figure, voice packets that enter the network at a constant rate can reach the intended destination by different routes. Because each of these routes may have different delay characteristics, the arrival rate of the packets may vary. This condition is called “jitter.” Another effect of multiple routes is that voice packets can arrive out of order. The voice- enabled router or gateway on the far end has to re-sort the packets and adjust the interpacket interval for a proper-sounding voice playout. 1-46 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Network transmission adds corruptive effects such as noise, delay, echo, jitter, and packet loss to the speech signal. VoIP is susceptible to these network behaviors, which can degrade the voice application. Ifa VoIP network is to provide the same quality that users have come to expect from traditional telephony services, then the network must ensure that the delay in transmitting a voice packet across the network and the associated jitter do not exceed specific thresholds. Packet Loss, Delay, and Jitter + Packet loss — Loss of packets severely degrades the voice application. + Delay ~ VoIP typically tolerates delays up to 150 ms before the quality of the call degrades. + Jitter ~ Instantaneous buffer use causes delay variation in the same voice stream. In traditional telephony networks, voice has a guaranteed delay across the network through strict bandwidth association with each voice stream, Configuring voice in a data network environment requires network services with minimal packet loss, low delay, and minimal jitter. Over the long term, packet loss, delay, and jitter will all affect overall voice quality. These voice quality problems are described here: = Packet loss: Your network will drop voice packets if the network quality is poor, if the network is congested, or if there is too much variable delay in the network. Coder-decoder (codec) algorithms can correct small amounts of loss, but too much loss can cause voice clipping and skips. The chief cause of packet loss is network congestion, = De End-to-end delay is the time that it takes the sending endpoint to send the packet to the receiving endpoint. End-to-end delay consists of the following two components: — Fixed network delay: You should examine fixed network delay during the initial design of the VoIP network. ITU standard G.114 states that a one-way delay budget of 150 ms is acceptable for high-quality voice. Research at Cisco Systems has shown that there is a negligible difference in voice-quality scores between networks built with 200-ms delay budgets and the public switched telephone network (PSTN). Examples of fixed network delay include propagation delay of signals between the sending and receiving endpoints, voice encoding delay, and voice packetization time for various VoIP codecs. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-47Example Variable network delay: Congested egress queues and serialization delays on network interfaces can cause variable packet delays. Serialization delay is a constant function of link speed and packet size. Larger packets combined with slower link- clocking speeds result in greater serialization delay. Although this ratio is known, can be considered variable because a larger data packet can enter the egress queue at any time before a voice packet. If the voice packet must wait for the data packet to serialize, the actual delay that is incurred by the voice packet is the sum of its own serialization delay plus the serialization delay of the data packet in front of it Jitter: Jitter is the variation between the time of the expected arrival of a packet and the time when it is actually received. VoIP endpoints use jitter buffers to tum the delay variations into a constant value so that voice can be played out smoothly. However, buffers can fill instantaneously, because network congestion can occur at any time within a network. This instantaneous buffer use can lead to a difference in delay times between packets in the same voice stream. When a calling party says, “Good morning, how are you?” the effect of packet loss, end-to-end delay, and jitter can be heard as follows = With packet loss, the called party hears, “Good m—ning, —w are you?” = With end-to-end delay, the called party hears, “... ... Good morning, how are you?” = With jiter, the called party hears, “Good morning, how are you?” 1-48 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Consistent Throughput + Throughput is the amount of data transmitted between two nodes in a given period. + Throughput is a function of bandwidth, error performance, congestion, and other factors. + Tools for enhanced voice throughput include: — Queuing ~ Congest ~ Header compression — RSVP — Fragmentation ‘Throughput is the actual amount of useful data that a source transmits to a destination. The amount of data that the originating end sends is nor necessarily the same amount of data that comes out at the destination. Error conditions in the network may affect the data stream. For example, bits may be corrupted in transit, leaving the packet unusable. Packets may also be dropped during times of congestion, potentially forcing a retransmit, using twice the amount of bandwidth for that packet. In the traditional telephony network, guaranteed bandwidth was associated with each voice stream, Cisco IOS software uses a number of techniques to reliably deliver real-time voice traffic across the modern data network. These techniques, which all work together to ensure consistent delivery and throughput of voice packets, are as follows: = Queuing: Queuing is the act of holding packets so that they can be handled with a specific priority when leaving the router interface. Queuing enables routers and switches to handle bursts of traffic, measure network congestion, prioritize traffic, and allocate bandwidth. Cisco routers offer several queuing mechanisms that can be implemented based on traffic requirements. Low latency queuing (LLQ) is one of the newest Cisco queuing mechanisms. = Congestion avoidance: Congestion-avoidance techniques monitor network traffic loads. ‘The goal is to anticipate and avoid congestion at common network and internetwork bottlenecks before it becomes a problem. These techniques provide preferential treatment in congested situations for premium-class (priority) traffic, such as voice traffic, These techniques also maximize network throughput and capacity use and minimize packet loss and delay, Weighted random early detection (WRED) is one of the QoS congestion avoidance mechanisms that the IOS software uses. = Header compression: In the IP environment, voice is carried in RTP, which is carried in User Datagram Protocol (UDP), which is then put inside an IP packet. This constitutes 40 bytes of an RTP/UDPIIP header. This header size is large when compared with the typical voice payload of 20 bytes, Compressed RTP (RTP) reduces the headers to 2 bytes in most cases, saving considerable bandwidth and providing better throughput. © 2006 Cisco Systems, Inc. ied CallManager Express Fundamentals 1-4= Resource Reservation Protocol (RSVP): RSVP is a transport layer protocol that enables a network to provide differentiated levels of service to specific flows of data. Unlike routing protocols, RSVP is designed to manage flows of data rather than to make decisions for each individual datagram, Data flows consist of discrete sessions between specific source and destination machines. Hosts use RSVP to request a QoS level from the network on behalf of an application data stream. Routers use RSVP to deliver QoS requests to other routers along the paths of the data stream. After an RSVP reservation is made, weighted fair queuing (WFQ) is the mechanism that actually delivers the queue space at each device Voice calls in the IP environment can request RSVP service to provide guaranteed bandwidth for a voice call in a congested environment. = Fragmentation: Fragmentation defines the maximum size for a data packet and is used in the voice environment to prevent excessive serialization delays. Serialization delay is the time that it takes to actually place the bits onto an interface. For example, a 1500-byte packet takes 187 ms to leave the router over a 64-kbps link. Ifa best-effort data packet of 1500 bytes is sent, then real-time voice packets are queued until the large data packet is transmitted, This delay is unacceptable for voice traffic. However, if best-effort data packets are fragmented into smaller pieces, then they can be interleaved with real-time voice packets. In this way, voice and data packets can be carried together on low-speed links without causing excessive delay to the real-time voice traffic. Reordering of Packets + IP assumes that packet-ordering problems exist. + RTP reorders packets. In traditional telephony networks, time-division multiplexing (TDM) carries voice samples in an orderly manner. The path is circuit-switched between the source and destination, so it is reserved for the duration of the call. All of the voice samples stay in order as they are transmitted across the wire. However, IP provides connectionless transport with the possibility of multiple paths between sites. This may cause voice packets to arrive out of order at the destination. Voice is carried in UDP IP packets, which do not provide automatic reordering of packets. RTP provides end-to-end delivery services for data that requires real-time support, such as interactive voice and video. According to RFC 1889, RTP provides these services: payload type identification, sequence numbering, time stamping, and delivery monitoring. 1-50 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Example In the figure, RTP reorders the voi ce packets using sequence numbers before playing them out to the user. The table illustrates the various stages of packet reordering by RTP. Sequencing of Packets by RTP Stage What Happens Voice packets enter the network. IP assumes that packet-ordering problems exist. RTP reorders the voice packets. The voice packets are put in order using sequence numbers, RTP retimes the voice packets. ‘The voice packets are spaced according to the time stamp that is contained in each RTP header. The user hears the voice packets in order and with the same timing as when the voice stream left the source. Real-Time Transport Control Protocol (RTCP) sends occasional report packets for delivery monitoring, Both the sender and receiver send occasional report packets containing information such as the number of packets sent or received, the octet count, and the number of lost packets. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-5Challenges in VoIP The traditional telephony network strives to provide the user with 99,999 percent uptime. This corresponds to 5.25 minutes per year of downtime. Many data networks cannot make the same claim. This topic describes methods that you can use to improve reliability and availability in data networks. Reliability and Availability + Traditional telephony networks claim 99.999 percent uptime. + Data networks must consider reliability and availability requirements when incorporating voice. + Methods for improving reliability and availability include: ~ Redundant hardware ~ Redundant links — UPS Proactive network management To provide telephony users the same level of service that they experience with traditional telephony, the reliability and availability of the data network take on new importance. When the data network goes down, it may not come back up for minutes or even hours. This delay is unacceptable for telephony users with network equipment such as voice-enabled routers, gateways, and switches for IP phones, because they will find that their connectivity is terminated. Administrators must provide an uninterruptible power supply (UPS) to these devices in addition to providing network availability. Previously, PSTN users received their power directly from the telephone company central office (CO) or through a UPS that was connected to their keyswitch or PBX in the event of a power outage. Now the network devices must have protected power to continue to function and provide power to the end devices. In traditional telephony, switches have multiple redundant connections to other switches. If either a link or a switch becomes unavailable, the telephone company can reroute the call. This design is why telephone companies claim a high availability rate High availability encompasses many areas of the network, and network reliability comes from incorporating redundancy into the network design. In a fully redundant network, you need to duplicate these components: m= Servers and call managers ™ Access layer devices, such as LAN switches = Distribution layer devices, such as routers or multilayer switches 1-52 IP Telephony Express (IPTX) v4.0 (© 2006 Cisco Systems, Inc= Core layer devices, such as multilayer switches = Interconnections, such as WAN links, even through different providers = Power supplies and UPSs In some data networks, a high level of availability and reliability is not critical enough to warrant the cost of the hardware and links required to provide complete redundancy. When you layer voice onto the network, you need to revisit the required level of availability and reliability. ‘The use of Cisco Unified CallManager clusters provides a way to design redundant hardware in the event of Cisco Unified CallManager failure. When using gatekeepers, you can configure backup devices as secondary gatekeepers in case the primary gatekeeper fails. When implementing redundancy, you must also revisit the network infrastructure. Redundant devices and Cisco IOS services, such as Hot Standby Router Protocol (HSRP), can provide high availability. For proactive network monitoring and trouble reporting, a network management platform such as CiscoWorks 2000 provides a high degree of responsiveness to network issues. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-53.Bandwidth Requirements in VoIP This topic describes the bandwidth that cach codec uses, and it illustrates the impact of the codec on total bandwidth as well as the effect of voice sample size on total bandwidth. This topic also lists overhead sizes for various Layer 2 protocols; it discusses how to use codecs, data links, and sample size to calculate the total bandwidth required for a VoIP call; and it describes the effect of voice activity detection (VAD) on total bandwidth. Bandwidth Implications of Codec 6.726 6.726 6.723 132 r16 153 eee] G.711 * | 64 | 32 16 5. 53 Laem] kbps | kbps kbps kbps * G.711 and G.729 are the only codecs supported by Cisco Unified CallManager Express. One of the most important factors for the network administrator to consider when building voice networks is proper capacity planning. Network administrators must understand how much bandwidth a VoIP call uses. With a thorough understanding of VoIP bandwidth, the network administrator can apply capacity-planning tools. Note Cisco IP phones support the G.711 and G.729 codecs. This is a list of codecs and their associated bandwidth: = The G.71I pulse code modulation (PCM) coding scheme uses the most bandwidth. It takes samples 8000 times per second, each of which is 8 bits in length, for a total of 64,000 bps. = The G.726 adaptive differential PCM (ADPCM) coding schemes use somewhat le bandwidth. Although each coding scheme takes samples 8000 times per second, as G.711 PCM does, it uses 4, 3, or 2 bits for each sample. The 4, 3, or 2 bits for each sample results in total bandwidths of 32,000 (G.726r32), 24,000 (G.726r24), or 16,000 bps (G.726r16), respectively. = The G.728 low-delay code excited linear prediction (LDCELP) coding scheme compresses PCM samples using codebook technology. It uses a total bandwidth of 16,000 bps. 1-54 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.@ = The G.729 and G.729A Conjugate Structure Algebraic Code Excited Linear Prediction (CS-ACELP) coding scheme compresses PCM using advanced codebook technology. It uses 8000 bps total bandwidth. = The G.723 and G.723A Multipulse Maximum Likelihood Quantization (MPMLQ) coding schemes use a look-ahead algorithm. These compression schemes result in 6300 (G.723r63) or 5300 bps (G.723r53), respectively. of a higher The network administrator should balance the need for voice quality against the c bandwidth in the network when choosing codecs. Higher codec bandwidth result cost for each call across the network. impact of Voice Samples ‘Couee em or G726a2 G73602 e726 e726r24 e726A6 G72606 e728 728 e729 e729 Grae G7a368 G723069 G77389 Voice sample size is a variable that can affect the total bandwidth that is used. A voice sample is defined as the digital output from a codec digital signal processor (DSP) that is encapsulated into a protocol data unit (PDU). Cisco uses DSPs that generate samples based on digitization of 10 ms worth of audio, Cisco voice equipment encapsulates 20 ms of audio in each PDU by default, regardless of the codec used. You can apply an optional configuration command to the dial peer to vary the number of samples encapsulated, Encapsulating more samples per PDU reduces the total bandwidth However, encapsulating more samples per PDU can result in larger PDUs, which ean cause variable delay and severe gaps if PDUs are dropped. Example Using the simple formula Bytes_per_Sample = (Sample_Size * Codec_Bandwidth) / 8, it is possible for you to determine the number of bytes encapsulated in a PDU based on the codec bandwidth and the sample size (20 ms is default). If you apply G.711 numbers, the formula reveals the following: = Bytes_per Sample = (,020 * 64,000) /8 = Bytes_per Sample = 160 © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals. 1-5!The figure illustrates various codecs and sample sizes and the number of packets that are required for VoIP to transmit 1 second of audio. Larger sample size results in larger packets and fewer encapsulated samples that have to be sent. This reduces bandwidth Data-Link Overhead + Ethernet: 18 bytes of overhead + MLP: 6 bytes of overhead + Frame Relay Forum 12 (FRF.12): 6 bytes of overhead Another contributing factor to bandwidth is the Layer 2 protocol that is used to transport VoIP. Alone and using uncompressed RTP, VoIP carries a 40-byte IP/UDP/RTP header. Depending on the Layer 2 protocol that is used, the overhead could grow substantially. As the Layer 2 overhead increases, the amount of bandwidth that is required to transport VoIP also increases. These points describe the Layer 2 overhead for various protocols = Ethernet: Carries 18 bytes of overhead. This total includes 6 bytes for source MAC address, 6 bytes for destination MAC address, 2 bytes for type, and 4 bytes for cyclic redundaney check (CRC). = Multilink PPP (MLP): Carries 6 bytes of overhead. This total includes I byte for flag, | byte for address, 2 bytes for control (or type), and 2 bytes for CRC. = Frame Relay Forum 12 (FRF.12): Carries 6 bytes of overhead. This total includes 2 bytes for data-link connection identifier (DLCI) header, 2 bytes for FRF.12, and 2 bytes for CRC. (FRF.12 is FRF.11 Annex C; FRF,11 is the implementation agreement for Voice over Frame Relay [VoFR].) 4-56 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Total Bandwidth Required era ao. Se ac Se .711_| 64000 | 240 | 76267 66133, 78933, G7 | 64000 | 160 | #2400 67200, "36400 G.726r32 | 32000 | 120 | 44267 34133 46933, G.726r32 | 32000 | 80 | 50400 35200 54400 G.726r24 | 24000 | 80 | 37800 "26400 “40800 G.726r24 | 24000 | 60 | 42400 27200, 48400 G.726r16 | 16000 | 80 | 25200 17600 27200 G.726r16 | 16000 | 49 | 34400 19200 38400 G.728_| 16000 | 80 _| 25200 17600, 27200 G.728 | 16000 | 40 | 34400 19200, 38400 G.728_| sooo | 40 | 17200 9600 19200 G.723_| sooo | 20 | 26400 11200 30400 G723r63 | 6300 | 48 | 12338 7350 13650 G.723r63 | 6300 18375, 8400 21000 G.723r53 | 5300 11395 6360 12720 6.723r53 | 5300 17490, 7420 20140 Codec choice, data-link overhead, sample size, and even cRTP all have positive and negative impacts on total bandwidth. To perform the calculations, you must have all of the contributing factors as part of the equation: = More required bandwidth for the codec equals more required total bandwidth = More overhead associated with the data link equals more required total bandwidth m= Larger sample size equals less required total bandwidth . tantly reduced required total bandwidth Example The formula Total_Bandwidth = ([Laver_2_Overhead + IP_UDP_RTP_Overhead + Sample Size] | Sample_Size) * Codec_Speed was used to produce the figure. For example, assume a G.729 codec and a 20-byte sample size using Frame Relay without RTP: = Total_Bandwidth = ([6 + 40 + 20] / 20) * 8000 = Total_Bandwidth = 26,400 bps © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals. 1-57Effect of VAD Femme Reiny | ame 76257 200 267 000 37800 25200 34400 25200, ‘34400 On average, an aggregate of 24 calls or more may contain 35 percent silence. With traditional telephony voice networks, all voice calls use 64-kbps fixed-bandwidth links regardless of how much of the call is conversation and how much is silence. With Cisco VoIP networks, all conversation and silence is packetized. VAD suppresses packets of silence. Instead of sending VolP packets of silence, VoIP gateways interleave data traffic with VoIP conversations to more effectively use network bandwidth. VAD is enabled by default for all VoIP calls. VAD provides a maximum of 35 percent bandwidth savings based on an average volume of more than 24 calls. Note Bandwidth savings of 35 percent is an average figure and does not take into account loud background sounds, differences in languages, and other factors. The savings are not realized on every individual voice call or on any specific point measurement. Note For the purposes of network design and bandwidth engineering, VAD should not be taken into account, especially on links that will carry fewer than 24 voice calls simultaneously. Various features, such as Music on Hold (MOH) and a fax function, render VAD ineffective. When the network is engineered for the full voice call bandwidth, all savings provided by VAD are available to data applications. Not only does VAD reduce the silence in VoIP conversations, but it also provides comfort noise generation (CNG). Because silence can be mistaken for a disconnected call, CNG provides locally generated white noise so that the call appears normally connected to both parties. 4-58 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Example The figure shows examples of the VAD effect in a Frame Relay VoIP environment. In the example using G.711 with a 160-byte payload, the bandwidth required is 82,400 bps. Turning VAD on reduces the bandwidth utilization to 53,560 bps. This results in a 35 percent reduction of bandwidth. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-!Summary This topic summarizes the key points that were discussed in this lesson. Tita «IP networks need to use QoS parameters and protocols to adequately support VoIP. + The characteristics of IP contribute to voice-traffic, problems, including packet loss, delay, and jitter. QoS mitigates delay, jitter, and packet loss in converged voice and data networks. + The codec, Layer 2 protocol, sample size, and VAD must all be taken into account when calculating VoIP bandwidth. VAD can lower bandwidth use as much as 35 percent, 1.80 IP Telephony Express (PTX) v4.0 (© 2006 Cisco Systems, nc.Lesson 4| Describing the Cisco Unified CallManager Express Voice Packet Handling Methods Overview This lesson describes the Cisco Unified CallManager Express voice packet handling methods It includes a discussion of IP phone calls, packet forwarding, priority and Real-Time Transport Protocol (RTP) stream information, and WAN call setup. Objectives Upon completing this lesson, you will be able to describe Cisco Unified CallManager Express voice packet handling methods. This ability includes being able to meet these objectives: = Describe the voice packet flow among various type of calls: calls between local IP phones (on-net calls), calls between IP phones and the PSTN (local calls), and calls from an IP phone to an IP phone over a WAN (intersite calls) = Describe voice packet forwarding, voice packet priority, and RTP stream information = Describe the requirements for setting up WAN calls, including DTMF relayIP Phone Calls This topic describes the process and steps for setting up a local (on-net) call. It describes these types of calls: = A call to the public switched telephone network (PSTN) using Cisco Uni Express as a PSTN gateway fied CallManager = A call to the PSTN using a separate PSTN gateway that is not the Cisco Unified CallManager Express router = A call flow that uses a WAN link to connect two IP phones registered to separate Cisco Unified CallManager Express routers On-Net Calls + SCCP is sent between IP phones and Cisco Unified CallManager Express. Cisco United + The voice connection is atttanager carried in IP packets SCoP messages” between two IP phones GIG) cnitcr pon foo, and has voice samples in an RTP segment. —_ / «there sno pera PU | gS°CP,. [Seo | loading on the Cisco nena ‘ a Unified CallManager a Express router except for —__ > call setup and teardown. By ——& 10.10.0.100:16922 RTP 10.10.0.101:18355 1000 The Cisco Unified CallManager Express system provides centralized call control for IP phones that register with the system. The system achieves this call control using the Skinny Client Control Protocol (SCCP, or Skinny). The IP phone uses SCCP after bootup to register with Cisco Unified CallManager Express. At this point, the IP phone cannot set up calls by itself and must send messages to Cisco Unified CallManager Express for even the simplest of actions. For example, when a caller lifts the handset off hook, Cisco Unified CallManager Express uses an SCCP message to instruct the IP phone to play a dial tone. When the call is connected, the IP phones use each other’s addresses to send the voice between the IP phones. Voice traffic is very delay-sensitive and drop-sensitive and does not withstand large jitter (variation in delay), so voice is carried in the form of data payloads inside RTP headers. RTP was designed to transport real-time traffic, such as voice. 4-62 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.The following illustrates the steps for completing a call from one local IP phone to another: Step 1 Step 2 Step 3 Step 4 Step 5 Step 6 Step 7 Step 8 An IP phone with extension 1000 goes off hook. Cisco Unified CallManager Express sends an SCCP message instructing phone 1000 to play a dial tone (which tells the caller that the system is ready to receive digits). The user on phone 1000 dials the digits “1-0-0-1.” As each digit is pressed, the phone sends an SCCP message to the Cisco Unified CallManager Express router, which analyzes the digits. (After the first digit, Cisco Unified CallManager Express sends an SCCP message telling the IP phone to stop playing the dial tone or, in some cases, to play a second dial tone.) A match is found to an IP phone with extension 1001 (phone 1001), and an SCCP ‘message is sent to the phone 1001 informing it of an incoming call. This message contains information about who is calling and instructions to phone 1001 to play the ring .wav file that is selected. Phone 1001 rings and is answered. The phone sends an SCCP message to Cis Unified CallManager Express that extension 1001 has been answered. Cisco Unified CallManager Express informs the IP phones that are involved with the call of the IP address, port, and coder-decoder (codec) that are to be used for the call The two IP phones set up RTP connections to each other, and the voice conversation can flow. Cisco Unified CallManager Express ceases to be involved in the call until the call is transferred or terminated. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals. 1-63PSTN Calls with Cisco Unified CallManager Express As the PSTN Gateway + SCCP signaling is used between the IP phone and “&Y. PSTN Cisco Unified CallManager 2 Express. Voice Analog or Digital Appropriate signaling is used Trunk(s) between Cisco Unified CallManager Express and the Cisco Unified PSTN. CalManager RTP is used to carry traffic Express between the IP phone andthe | Cisco Unified CallManager Voice over IP Express router. UDP 16,384— Signaling 32,768 Cisco Unified CallManager wy TCP 2000 Express acts as an MTP. + Voice is sent to the PSTN. When calls are made to or from the PSTN and are coming from or destined for an IP phone that is under the control of Cisco Unified CallManager Express, the RTP stream must terminate on a Media Termination Point (MTP). The call must then be converted to the format that is appropriate for the type of trunk that is going to the PSTN. The following illustrates the steps for completing a call from onc local IP phone to a PSTN destination with the Cisco Unified CallManager Express router acting as the PSTN gateway: Step1 An IP phone with extension 1000 goes off hook for the 1000 extension. Step2 Cisco Unified CallManager Express sends an SCCP message instructing phone 1000 to play a dial tone (which tells the caller that the system is ready to receive digits). Step 3 The user on phone 1000 dials the digits of the PSTN destination. As each digit is, pressed, the phone sends an SCCP message to the Cisco Unified CallManager Express router, which analyzes the digits. (After the first digit, Cisco Uni CallManager Express sends an SCCP message telling the IP phone to stop playing the dial tone or, in some cases, to play a second dial tone.) Step 4 A match is found to the PSTN destination, and the Cisco Unified CallManager Express router (which in this case is the PSTN gateway) seizes a trunk, either analog or digital Step5 When the call is connected from the PSTN, an RTP stream is set up between the IP phone and the PSTN gateway. The RTP stream acts as an MTP. The voice inside the RTP stream is converted to the format of the trunk across which the voice goes. -64 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.PSTN Calls with a Separate Voice Gateway SCCP signaling is used between the IP phone and Cisco Unified Cisco CallManager Express. Analog or Unified H.323 is used between camanager Cisco Unified CallManager Trunk(s) Express and the PSTN gateway. > RTP is used to carry traffic H.323 between the IP phone and RTP a the voice gateway. . The voice gateway acts as an MTP. Voice is sent to the PSTN &y from the voice gateway. SccP Signaling When a caller places a call to or from the PSTN that are coming from or destined for an IP phone that is under the control of Cisco Unified CallManager Express, the RTP stream must terminate on an MTP. The following illustrates the steps for completing a call from one local IP phone to a PSTN destination when the Cisco Unified CallManager Express system is not the PSTN gateway. Step1 An IP phone with extension 1000 goes off hook for the 1000 extension. Step2 The Cisco Unified CallManager Express system sends an SCCP message instructing phone 1000 to play a dial tone (which tells the caller that the system is ready to receive digits). Step3 The user on phone 1000 dials the digits of the PSTN destination. As each digit is pressed, the phone sends an SCCP message to the Cisco Unified CallManager Express router, which analyzes the digits. (After the first digit, Cisco Unified CallManager Express sends an SCCP message telling the IP phone to stop playing the dial tone or, in some cases, to play a second dial tone.) Step4 A match is found to the PSTN destination. Step5 Because Cisco Unified CallManager Express does not physically terminate the trunk to the PSTN terminated locally, it must signal the PSTN gateway to set up a connection to the IP phone. The call control protocol, either H.323 or session initiation protocol (SIP), must be used to set up the call. Step6 On the PSTN gateway trunk, the system can use cither analog or digital to connect to the PSTN Step7 The IP phone and the PSTN gateway set up an RTP session. The RTP stream is converted to the format that the PSTN connection uses. Step8 The Cisco Unified CallManager Express router ceases its involvement until the call is transferred or terminated. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-65Intersite Calls + SCCP signaling is used between the IP phone and Cisco Unified CallManager Express. + H.323 or SIP signaling is used between the Cisco Unified CallManager Express routers. + RTP is used to carry traffic between the IP phones. ~ IfVolP is used on the WAN, the RTP header will be preserved. - The following illustrates the steps for completing a call that starts from an IP phone that is under the control of one Cisco Unified CallManager Express router and goes across a WAN link to an IP phone that is controlled by another Cisco Unified CallManager Express router. Step 1 Step 2 Step 3 Step 4 Step 5 Step7 Step 8 An IP phone with extension 1000 goes off hook. Cisco Unified CallManager Express sends an SCCP message instrueting phone 1000 to play a dial tone (which tells the caller that the system is ready to receive digits) The user on phone 1000 dials the digits “2-0-0-0." As each digit is pressed, the phone sends an SCCP message to the Cisco Unified CallManager Express router, which analyzes the digits. (After the first digit, Cisco Unified CallManager Express sends an SCCP message telling the IP phone to stop playing the dial tone or, in some cases, to play a second dial tone.) A match is found to the dialed number, 2000, across the WAN link. Cisco Unified CallManager Express uses the voice gateway function (in this case, the Cisco Unified CallManager Express router is the voice gateway) to set up a call to the remote Cisco Unified CallManager Express system, Either H.323 or SIP i used to set up this call When the remote Cisco Unified CallManager Express system receives the call setup message for extension 2000, the system sends an SCCP message to the IP phone with extension 2000, causing it to ring. ‘When phone 2000 is answered, an SCCP message goes from its Cisco Unified CallManager Express router to the IP phone to which system that the IP phone answered the call Via either H. or SIP, the remote Cisco Unified CallManager Express router sends a message that the call has been answered, The message is sent to the Cisco Unified CallManager Express router with which phone 1000 is associated. 66 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc.Step9 In this case, because the Cisco Unified CallManager Express routers are the voice gateways, the RTP packets traverse the routers. (However, to the routers, the RTP packets are just data.) The Cisco Unified CallManager Express router ceases to be involved in call control until the call is transferred or terminated. Note ‘As long as the path across the WAN link is entirely IP-based, the RTP header will be preserved. © 2006 Cisco Systems. Inc. Cisco Unified CallManager Express FundamentalsPacket Forwarding, Voice Packet Priority, and RTP Stream Information This topic describes the quality of service (QoS) markings, cost of service (CoS), and IP precedence that the IP phone places in voice packets at Layer 2 and Layer 3, respectively. The topic also describes the concept of voice encapsulation Cisco Unified CallManager Express Local QoS A call has QoS markings on the Layer 2 header and in the IP packet header. Layer 2.Cos. Layer 31P Marking of 5 Precedence Marking of 5 + QoS markings are used to give voice traffic priority over most other types of data on the network. + The Cisco Unified CallManager Express system requires all IP phones under its contro! to be local on the same LAN network. When voice is generated and put into IP packets on an IP phone, both Layer 2 and Layer 3 QoS markings are present. The Layer 2 marking is present only if the connection to the IP phone is an 802.1Q trunk. An 802.1Q trunk is the recommended configuration. The Layer 2 QoS marking is called “CoS.” CoS has a range of 0 through 7, with 7 being the highest priority When an IP phone uses an 802.1Q Ethernet header to generate IP voice packets, the CoS marking is set to the value of 5 by default. This marking allows the switch to give preferential treatment to voice frames There is an IP precedence marking in the Layer 3 IP header, which also has a range of 0 through 7. The IP precedence marking is also set to 5 by default for voice that is generated on the IP phone. 1-68 _IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IncRTP Stream Information mee RTP Voice Payload RTP + RTP headers carry voice across an IP-based network. + The RTP header is carried inside a UDP segment. + The UDP segment is carried inside IP packets. + UDP ports are randomly selected from 16,384 through 32,768. + Ifthe whole path is VoIP, the RTP header will be preserved. 6 Sn nm Voice that is generated on an IP phone is carried inside an RTP header. The RTP header is encapsulated inside a User Datagram Protocol (UDP) segment, The UDP segment has a randomly selected port for the current conversation, which will be in the range of 16.384 through 32,768. This UDP segment is then encapsulated inside an IP packet with an IP precedence marking of 5. The IP packet is then put into an Ethernet frame and sent to the attached switch. The RTP header will be unchanged as the long as the path is an all-IP-based network. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 16WAN Call Setup This topic explains the need for Call Admission Control (CAC) and describes what CAC is. It also explains the need for dual tone multifrequency (DTMF) relay over a WAN. The Need for Call Admission Control CAC is useful for the WAN environment, where band h is often limited. Is there enough bandwidth on the WAN for three simultaneous calls? + If allowed, the third call will cause quality problems not only for the third call, but also for all three calls. + The third call should be prevented. eee When calls are to be sent across an IP WAN link, saturation of the bandwidth is possible. When there is not enough bandwidth, the effect on voice conversations can be significant. Packets are dropped or queued on the interface, which results in a significant degradation of service. Insufficient bandwidth may result when voice traffic is sharing the link with other types of data Insufficient bandwidth may be managed using QoS tools. When these tools are being used, preference should be given to voice traffic. In addition, degradation of service results from too much voice traffic on a link, which can cause all calls to receive poor quality. For example, in the figure, it is assumed that there is enough bandwidth for two simultaneous calls. Ifa third call is allowed to use the WAN, that third call and the other two calls will suffer from choppy audio. The best practice is to prevent the third call from using the link. To limit the number of calls across a WAN link, a CAC mechanism is needed. This CAC mechanism can be set up to allow only a certain number of calls on a WAN link. 4-70 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Call Admission Control Locally CAC is not needed for traffic to IP phones because Cisco Unified CallManager Express assumes that the medium is Ethernet LAN and therefore that the bandwidth is, effectively, unlimited. There is no need for a CAC mechanism locally between the IP phones and Cisco Unified CallManager Express because all IP phones under the control of Cisco Unified CallManager Express must connect via a LAN to the Cisco Unified CallManager Express router. The mucl larger amount of bandwidth on an Ethernet LAN negates the need for a CAC mechanism. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express FundamentalsCall Admission Control Across WANs CAC should be used for WAN links that could be even temporarily saturated. CAC is implemented through an H.323 mechanism called a gatekeeper. The voice gateway asks the gatekeeper if there is enough bandwidth to set up the call with a specific codec. The gatekeeper answers the question with either an affirmative or a negative response. If the answer is negative, the dial plan of the voice gateway must either connect the call using a secondary path, such as. the PSTN, or give a fast busy signal to the caller. Over a WAN, the CAC mechanism is usually implemented through an H.323 mechanism called a “gatekeeper.” The gatekeeper is consulted by the voice gateway, which in many cases is the Cisco Unified CallManager Express router, to determine whether sufficient bandwidth is available for the call to be set up. The gatekeeper, which has been configured to allow a certain amount of bandwidth to be available for voice, responds affirmatively or negatively. If the answer is affirmative, the voice gateway sets up the call. If the answer is negative, the voice gateway either looks for alternate ways to get to the destination or plays a fast busy signal The use of a gatekeeper ensures that no more than a certain amount of bandwidth is consumed by voice traffic on a WAN. Tip A gatekeeper is used for other functions as well. For more information, go to http:www. cisco.com/en/US/techitk1077/technologies_tech_note09186a00800a8928. shtml 1-72 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, InceR eee Mee tm datn WT | DTMF tones are normally carried in-band with voice. Low-bandwidth codecs such as G.729 are designed for human voice, not for DTMF tones, and they can distort DTMF tones carried in-band. ‘Symptoms of this problem are DTMF tones that are interpreted as another digit or not detected at all. The solution is to send DTMF tones out-of-band in packets or in-band with a special RTP packet. Various types of DTMF relay mechanisms exist. When calls are sent across a slower WAN link, low-bandwidth codecs are often used to conserve bandwidth. These low-bandwidth codecs can have problems carrying DTMF digits. The DTMF digits can be misinterpreted or not seen as valid tones when carried in-band with voice. The G.729 codec is especially susceptible to these problems. The problems can show up when voice mail is being checked and when interactive voice response (IVR) is being used. Because of the problems arising from the use of low-bandwidth codecs, the DTMF digits should be carried out-of-band from the voice. The IP phones in the Cisco Unified CallManager Express system already use DTMF relay by using SCCP when a digit is pressed on an IP phone during call setup. After the call is dialed, the DTMF relay and whether it will be used across a WAN link are defined on the voice gateway, Note Ifthe G.711 codec is used everywhere, DTMF relay is not required, although implementing it is still recommended. There is no adverse effect of implementing DTMF relay. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals. 1-73Summary This topic summarizes the key points that were discussed in this lesson. + Local calls are set up and torn down by Cisco Unified CallManager Express, but RTP goes between the two IP phones. SCP is used between the IP phones and Cisco Unified CallManager Express. Calls to the PSTN can use the Cisco Unified CallManager Express router as the gateway or as a separate router. The PSTN gateway must act as an MTP and convert the RTP ‘stream to and from the format of the connection to the PSTN. + Intersite calls that use an IP WAN link between sites preserve the RTP headers. CAC should be used when going across low-bandwidth WAN links, and DTMF relay should be used. when low-bandwidth codecs are used across WAN links. 4-74 IP Telephony Express (PTX) v4.0 (© 2006 Cisco Systems, neModule Summary This topic summarizes the key points that were discussed in this module. Module Summary Cisco Unified CallManager Express provides the small and medium-sized business with an integrated solution for call control, voice mail, and data services. Voice may be placed as data in packets through a process of sampling the voice, quantizing the samples, and encoding the value as a binary expression. Packet loss, delay, jitter, and the required bandwidth all must be considered when configuring VoIP. Cisco Unified CallManager Express sets up calls through the use of protocols such as SCCP, RTP, H.323, and SIP. When you are moving from a traditional telephony environment to a VoIP environment, it is important to understand the differences and similarities between the two. The VoIP process takes voice samples and represents them as data, which is then collected into samples that are put into Real-Time Transport Protocol (RTP) segments. The RTP segments are placed into User Datagram Protocol (UDP) segments, then into IP packets. Finally, the IP packets are placed into Ethernet frames and carried across the network. You need to understand the challenges that you will encounter in the data environment when you are designing and deploying Cisco Unified CallManager Express. The challenges include delay, jitter, packet loss, knowing the required bandwidth, and the need to give preference to VolP packets. You must be able to solve these challenges with the many Cisco IOS tools built into Cisco Unified CallManager Express. An understanding of the basic call flows of Cisco Unified CallManager Express is also intial to understanding the issues and challenges. One of the most challenging situations is sending VoIP across an IP WAN link to another site. Many issues arise when WAN links are involved. These include bandwidth, Call Admission Control (CAC), quality of service (QoS), and others. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-References For additional information, refer to these resources: = Cisco Systems, Inc. Cisco Unified CallManager Express—IP Communications for the Small or Branch Office. http://www cisco.com/en/US/products/sw/voicesw/ps4625/prod_brochure09186a00801c63 68.html. = Cisco Systems, Ine. Cisco Unity Express—IP Communications for the Small or Branch Office. http://www cisco.com/en/US/products/sw/voicesw/ps5520/prod_brochure09186a00801.c63 68.html, m= Cisco Systems, Ine. Voice over IP—Per-Call Bandwidth Consumption. http://www cisco.com/warp/public/788/pkt-voice-general/bwidth_consume.html#related. Cisco Systems, Inc. Voice Quality—Introduction http://www cisco.com/en/US/tech/tk652/tk698/tsd_technology_support_protocol_home.ht ml. = Cisco Systems, Ine. Voice Quality—Quality of Service for Voice over IP. http://www.cisco.convVen/US/tech/tk652/tk698/technologies_white_paper09186a00800d6b 73.shtml. ™ Cisco Systems, Inc, Cisco Unified CallManager Express—Configuration Guides. htip://vww.cisco.com/en/US/products/sw/voicesw/ps4625/products_installation_and_confi guration_guides_list.html 1-76 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Module Self-Check Use the questions here to review what you learned in this module. The correct answers and solutions are found in the Module Self-Check Answer Key Qh Q2) Q3) Q4) Which best describes Cisco Unified CallManager Express? (Source: Describing Key Features of Cisco Unified CallManager Express and Cisco Unity Express) A) anoptional feature of Cisco IOS software that supports up to 240 users B) _ a standard feature of Cisco IOS software that supports up to 240 users C) _ anoptional feature of Cisco 10S software that supports up to 120 users D) _astandard feature of Cisco IOS software that supports up to 120 users Which three Cisco router series include Cisco Unified CallManager Express available on Cisco IOS software? (Choose three.) (Source: Describing Key Features of Cisco Unified CallManager Express and Cisco Unity Express) A) 3700 B) 2600 c) 3800 D) 1600 E) 1700 Which best describes Cisco Unity Express? (Source: Describing Key Features of Cisco Unified CallManager Express and Cisco Unity Express) A) available as a software upgrade B) available in a network module that supports up to 8 hours of voice message storage C) available in a network module that supports up to 20 hours of voice message storage D) available in an advanced integration module that supports up to 14 hours of voice message storage Which feature set does Cisco Unity Express include? (Source: Describing Key Features of Cisco Unified CallManager Express and Cisco Unity Express) A) voice mail and automated attendant for large enterprise offices B) two call control options: Cisco Unified CallManager and Cisco Unified CallManager Express C) complete integration into Cisco 2600, 3600, and 3700 Series routers D) three form factors: software upgrade, network module, and advanced integration module What defines how many phones will be controlled with the CallManager Express software? (Source: Describing Key Features of Cisco Unified CallManager Express and Cisco Unity Express) A) feature license B) specific Cisco Unified CallManager Express-enabled Cisco IOS image license C) seat license D) Cisco Unity Express license ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-77Q6) Which mailbox license size is not available for the AIM-CUE? ((Source: Describing Key Features of Cisco Unified CallManager Express and Cisco Unity Express) A) 2B B) 25 oC 50 D) 180 Q7)__ Which is used by Cisco Unified CallManager Express to provide call processing for IP phones? (Source: Describing Key Features of Cisco Unified CallManager Express and Cisco Unity Express) A) RTP B) H323 C) PSTN D) —-sccP 8) Match the component of a telephony network with the function that it performs. (Source: Explaining Differences Between Traditional Telephony and VoIP) A) private or CO switch B) edge device Cc) trunk D) local loop 1. handles signaling, call routing, call setup, and call teardown 2. _ provides a path between two switches 3. connects to the PSTN 4. _ interfaces to the telephone company network 9) Which step is optional in analog-to-digital conversion? (Source: Explaining Differences Between Traditional Telephony and VolP) A) compression B) encoding C) quantization D) sampling Q10) Which two coding schemes are examples of waveform algorithms? (Choose two.) (Source: Explaining Differences Between Traditional Telephony and VoIP) A) PCM B) ADPCM Cc) CELP D) —_LDCELP E) (CS-ACELP QI1) To what size does RTP compress the IP/UDP/RTP header without using a UDP checksum? (Source: Explaining Differences Between Traditional Telephony and VoIP) A) 2bytes B) 4 bytes C) 8 bytes D) 12 bytes -78 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Qi) Q13) Qi4) QIS) Q16) Which two factors have a minimal effect on data transmissions but negatively impact voice transmissions? (Choose two.) (Source: Understanding VoIP Challenges and Solutions) A) high bandwidth B) TI links C) packet loss D) _ jitter E) Layer 2 protocol Which two Cisco IOS QoS features are employed in the output queue of the router? (Choose two.) (Source: Understanding VoIP Challenges and Solutions) A) FRF.I2 B) IP to ATM CoS Cc) CBWFQ D) —cRTP E) RSVP F) — WRED Which two Cisco QoS features are deployed in a WAN? (Choose two.) (Source: Understanding VoIP Challenges and Solutions) A) CAR B) — DWFQ ©) MLP with LFI D) —_ QoS policy propagation via BGP E) — cRTP Which coding scheme requires the least bandwidth with RTP applied? (Source: Understanding VoIP Challenges and Solutions) A) GTI B) — G.723, Cc) G.726 D) = -G.729 In which two call scenarios do the RTP packets, after the call is set up, continue to traverse the CallManager Express router or routers for the remainder of the call until it is transferred or terminated? (Choose two.) (Source: Describing the Cisco Unified CallManager Express Voice Packet Handling Methods) A) local (on-net) calls B) _ acall to the PSTN using Cisco Unified CallManager Express as a PSTN gateway C) call to the PSTN using a separate PSTN gateway that is not the CallManager Express router D) _acall flow using a WAN link to connect two IP phones registered to separate Cisco Unified CallManager Express routers that are acting as the voice gateways © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-79Qiy) QI8) QI9) Q20) In which call scenario does the voice gateway act as an MTP? (Source: Describing the Cisco Unified CallManager Express Voice Packet Handling Methods) A) a call between an IP phone and the PSTN (local call) B) a call between local IP phones (on-net call) C) __ acall using a WAN link to connect two IP phones that are registered to separate Cisco Unified CallManager Express routers D) call using a Cisco ATA in conjunction with a CUE module Which is Layer 2 marking? (Source: Describing the Cisco Unified CallManager Express Voice Packet Handling Methods) A) 802.1Q B) QoS © CoS D) CAC In which is an RTP header encapsulated? (Sou escribing the Cisco Unified CallManager Express Voice Packet Handling Methods) A) aTCP segment B) a UDP segment C) either a TCP segment or a UDP segment, depending on which is supported by the network D) a TCP, UDP, or frame to packet segment Which call scenario is most likely to require CAC? (Source: Describing the Cisco Unified CallManager Express Voice Packet Handling Methods) A) alocal (on-net) call B) a call to the PSTN using Cisco Unified CallManager Express as a PSTN. gateway C) _ acall to the PSTN using a separate PSTN gateway that is not the Cisco Unified CallManager Express router D) _acall flow using a WAN link to connect two IP phones that are registered to separate Cisco Unified CallManager Express routers that are acting as the voice gateways -80 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Module Self-Check Answer Key Qn A @) — A.B.C Q3) D Q4) Q5) Q6) Q7) Q8) Q9) 10) Qi Q12) QI) Ql4y Qis) QI6) Qiyy QI8) Ql9) 020) PpuUors a F o w veEn>srnnas es >>> © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Fundamentals 1-81Module 2| Cisco Unified CallManager Express Configuration Overview This module describes the basic functionality of Cisco Unified CallManager Express. This information includes the configuration of specific network components and services that are necessary for the proper functioning of Cisco Unified CallManager Express. This module also discusses the files that are required to run the phones and the web-based GUL Module Objectives Upon completing this module, you will be able to describe the features and functionality of Cisco Unified CallManager Express. You also will be able to configure Cisco Unified CallManager Express to support IP phones. This ability includes being able to meet these objectives: = Describe the key features and functionality of Cisco Unified CallManager Express = Configure Cisco Unified CallManager Express network parameters and discuss the need for and configuration of auxiliary VLANs, DHCP, DHCP relay, and NTP- Describe the IP phone registration process Define the SCCP IP phone and instances and types using ephone and ephone-dn Load the SIP firmware and configure the SIP IP phone Describe Cisco Unified CallManager Express files Describe the three ways to create an initial phone setupLesson 1 Understanding Cisco Unified CallManager Express Features and Functionality Overview This lesson introduces you to the key features and function: Express y of Cisco Unified CallManager Objectives Upon completing this lesson, you will be able to describe the key features and functionality of Cisco Unified CallManager Express. This ability includes being able to meet these objectives: m= Identify the key benefits and features of Cisco Unified CallManager Express = Describe the supported platforms and telephones for Cisco Unified CallManager Express = Describe the supported protocols and integration options for Cisco Unified CallManager Express m= Describe Cisco Unified CallManager Express requirements for licensing, memory, platforms, Cisco IP phone models, and software = Identify Cisco Unified CallManager Express restrictionsKey Benefits and Features This topic describes the key benefits and features of Cisco Unified CallManager Express, Cisco Unified CallManager Express Gaal Extends capabilities to the small office that were previously available to larger enterprises only + Reduces TCO by delivering voice, video, and data over a consolidated infrastructure Based on Cisco IOS software Supports converged applications Protects customer investment Administered by GUI or CLI IP telephony is currently undergoing explosive growth, driven by access to value-added features and applications that only IP telephony can provide to the end user. This growth allows you to extend the Cisco Unified CallManager Express benefits and features to the small office In addition, the cost benefits of converging voice, video, and data onto a single network are fueling the rapid acceptance of IP telephony. The reduction in the total cost of ownership (TCO) is one of the main benefits of the Cisco Unified CallManager Express solution. Because the solution is based on Cisco IOS software, you can leverage your experience with Cisco products to offer simple configuration and deployment. You can integrate Cisco Unified CallManager Express into a multiservice router, allowing you to take advantage of converged applications, including content networking, video, quality of service (QoS), firewall, Ethernet, and Extensible Markup Language (XML) services. The Cisco Unified CallManager Express solution provides investment protection for customers when they need to migrate to a centralized Cisco Unified CallManager architecture. Administration and management are accomplished through either the familiar Cisco IOS software command-line interface (CLI) or a web-based GUI. 2-4 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Cisco Unified CallManager Express Key Features + Phone features + System features + Trunk features + Voice mail features Cis .0 Unified CallManager Express has many high-level phone, system, trunk, and voice mail features. Phone Features The high-level phone features for Cisco Unified CallManager Express are as follows: Support for single-line and multiline Cisco IP phones (Cisco Unified IP Phone 7902G, 7905G, 7910G+SW, 7912G, 7920, 7940G, 7941G, 7960G, 7961G, 7970G, and 7971G-GE models) Support for the Cisco Unified IP Conference Station 7935 and 7936 Support for analog phones on the Cisco Unified CallManager Express router analog voice ports and on the Cisco ATA 186 and 188 Analog Telephone Adaptor models Support for fax machines XML services on Cisco IP phones 240 phones per system Six line appearances per each Cisco Unified IP Phone 7960G or 7961G Eight line appearances per each Cisco Unified IP Phone 7970G and 7971G-GE On-hook dialing Local directory lookup Speed dial and last-number redial Idle URL, which can periodically push messages onto the screen of Cisco Unified IP Phone 7940G, 7941G, 7960G, 7961G, 7970G, or 7971G models Auto attendant functionality when the Cisco Unified IP Phone 7940G 7941G, 7960G, or 7961G phone is combined with the Cisco Unified IP Phone Expansion Module 7914 © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2,Configurable ring types Message Waiting Indicator (MW1) Customization of softkeys Do Not Disturb (DND) feature to divert calls directly to voice mail IP phone display of DND state Enable and disable call waiting notification per line = Monitor-line button speed dial System Features The high-level system features for Cisco Unified CallManager Express are as follows: \ = Conferencing capabilities — no lait Por Quabar of Pechaperts = Paging @ Intercom ® Call transfer—consultative and blind ® Call hold and call retrieve = Call pickup of on-hold calls = Call waiting = = Tone on hold and tone on transfer for internal calls = Music on hold (MOH) and music on transfer for external calls = MOH—file on router = MOH live feed—external source = Distinctive ringing —internal versus external = International language support—German, French, Italian, and Spanish m= System speed dial option via XML service = Directory services using XML = Web-based GUI for moves, adds, and changes = GUI customization capabilities m Interactive voice response (IVR) Auto Attendant = Class of restriction to restrict calling capabilities = Inline power for IP phones = Call transfer and call forwarding (standards-based—H450.2 and H450.3) = Computer telephony integration (CTI) support with Telephony Application Programming Interface (TAP!) Lite ® Call Detail Record (CDR) generation via RADIUS. = Interworking with Cisco and NetCentrex gatekeepers = Hookflash pass-through to a central office (CO) for analog phones = Date and time synchronization with Network Time Protocol (NTP) 2-6 IP Telephony Express (IPTX) v4.0 © 2008 Cisco Systems, Inc.= Longest-idle hunt group = Hunt group dynamic login and logout = Hunt group statistics = Caller ID display for hunt group = Called name directory lookup for Dialed Number Identification Service (DNIS) = Called name display for overlay dialed number (DN) = Conference initiator drop-off = Consultative transfer for direct station select = Repeat night service notification every 12 seconds = Translation-profile support for Ethernet phone directory number (ephone-dn) Trunk Features The high-level trunk features for Cisco Unified CallManager Express are as follows: ® Direct inward dialing (DID) and direct outward dialing (DOD) = BRI and PRI support—all switch types that Cisco 1OS software supports = Caller identification display and blocking, calling name display. and automatic number identification support Analog—Foreign Exchange Office (FXO), DID Digital trunk support—T1 and El WAN link support—Frame Relay, ATM, Multilink PPP (MLP), and DSL Network calls using H.323 Dedicated trunk mapping to phone button H.323 to session initiation protocol (SIP) call routing to Cisco Unity Express RFC 2833 support over SIP trunks Transcoding e-Mail Features The high-level voice-mail features for Cisco Unified CallManager Express are as follows: m= Integration with Cisco Unity voice mail = Integration with Cisco Unity Express voice mail = Third-party voice-mail integration—H.323, analog dual tone multifrequency (DTMF) Tip You can find the Cisco Unified CallManager Express System Administrator Guide at http:/wmw.cisco.comlen/US/products/sw/voicesw/ps4625/products_configuration_guide_bo 0k09186a00806a80d¢.html © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2+Supported Platforms and Telephones This topic describes the supported platforms and telephones of Cisco Unified CallManager Express. Supported Platforms Cisco Unified CallManager Express supports these Cisco platforms: JAD 2430 Series (SP only) 1751-V + 1760 2610XM 2611XM 2620XM 2621XM 2650XM 2651XM 2691 + 2801 + 2811 2821 + 2851 3725 + 3745 3825 3845 ee Cisco Unified CallManager Express supports these Cisco platforms: CiscolAD 2430 Integrated Access Device (SP only); Cisco 1751-V and 1760 Modular Access Routers; Cisco 2610XM, 2611XM, 2620XM, 2621XM, 2650XM, and 2651XM Multiservice Routers; Cisco 2691 Multiservice Platform; Cisco 2801, 2811, 2821 Routers; and Cisco 3725 and 3745 Multiservice Access Routers. , 3825, and 3845 Integrated Services 28 IP Telephony Express (IPTX) v4.0. ‘© 2006 Cisco Systems, IncSupported Platforms (Cont.) ee) Tena Number of faced TAD 2430, 1751 24 FLCOMESMALL 72610XM, 2611xM, 2620xM, | 36 Faoue 2621XM, 2811 ‘2650XM, 2651XM, 2621 48 To eea mab 2697 72 Teena 26 Tae 7144 err Depending on the platform, Cisco Unified CallManager Express supports up to 24, 36, 48, 72, 96, 144, 168, 192, or 240 IP phones. You can purchase the licenses and upgrade them incrementally, allowing you to purchase only the required number of licenses now, with the ability to grow in the future by purchasing additional licenses. Example ACME Company currently has an installation of 72 IP phones, with each employee having an IP phone. ACME has also purchased a Cisco 3745 Multiservice Access Router because it plans to hire 38 additional employees in the next year, for a total of 110 employees. Each employee will need to have an IP phone. Initially, ACME purchased the feature license FL-CCME-96, which is the minimum license required to support 72 IP phones. When the expansion to 110 IP phones becomes necessary, ACME can purchase the feature license FL-CCME-SMALL to add 24 IP phones to the Cisco Unified CallManager Express system. The two licenses together will allow up to 120 IP phones. which will support the planned expansion to 110 IP phones. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-Supported Cisco IOS Images + Cisco Unified CallManager Express Release 4.0 requires a minimum of Cisco IOS Release 12.4 (9)T. + The version of Cisco IOS Release 12.4 (9)T must contain the IP voice feature set for all supported platforms except the Cisco 1700 Series Modular Access Router. + The Cisco 1700 Series Modular Access Router must have the VOX feature set of Cisco IOS Release 12.4 (9)T. Cisco Unified CallManager Express Release 4.0 requires a minimum of Cisco IOS Software Release 12.4(9)T. The Cisco IOS release must also include the IP voice feature set to include the CallManager Express functionality. When you are using Cisco 1700 Series Modular Access Routers, the version of Cisco IOS software that is required is Cisco IOS Release 12.4(9)T. and it must contain the VOX feature set 10 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Cisco Unified CallManager Express Scalability and Memory Requirements + Memory recommendations may not be sufficient for larger installations. + The memory that is needed depends upon: The applications that are configured ~The hardware platform Memory requirements for the Cisco Unified CallManager Express router depend on the number of IP phones and the applications that you configure on the router. For example, if Network Address Translation (NAT) is also running on the router, the memory requirements will be greater than if only Cisco Unified CallManager Express is running on the router. The memory that you need to install in the router varies based on the hardware platform and is one factor that determines the number of IP phones that the Cisco Unified CallManager Express router will support. The table breaks down the capabilities of Cisco IOS Software Release 12.4(9)T with Cisc Unified CallManager Express 4.0 when deployed on various Cisco router platforms, © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-1Cisco IOS Software Release 12.4(9)T with Cisco Unified CallManager Express 4.0 Platform Phones | Extensions or imum Directory Numbers | Recommended Flash/RAM. 1AD2430, 1760, 1760-V_ | 24 120 64/128 1751-V 24 120 32/128 ‘2610XM, 2611XM, 36 144 48/128 2620XM, 2621XM, 2811 2650XM, 2651XM 48 192 48/128 2691 72 288 64/254 2801 24 120 64/128 2821 48 144 63/256 2851 96 288 64/254 3725 144 500 64/256 3745 192 500 64/256 3825 168 500 64/256 240 720 65/256 3845 42 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Supported Telephones (eB ee > & Yaw 7940 + 7914, 7971G-GE 7960+ 7914 ATA 186, 188, Cisco Unified CallManager Express supports the new generation of intelligent Cisco Unified IP Phones, including the 7902G, 7905G, 7910+SW, 7911G, 7912G, 7920, 7940G, 7960G, 7941G, 7941G-GE, 7961G, 7961G-GE, 7970G, 7971G-GE, 7940G + 7914, and 7960G +7914 phones, and the 7935 and 7936 conference stations. Regular analog phones and fax machines are supported through the Cisco ATA 186 and 188 Analog Telephone Adaptors or Foreign Exchange Station (FXS) ports on the Cisco Unified CallManager Express router. All supported telephones use Skinny Client Control Protocol (SCCP, or Skinny). © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-13.7902G Features + Common-area phone * G.711 and G.729 codecs Single line No display SCCP support + Four programmable keys + Prestandard Power over Ethernet The Cisco Unified IP Phone 7902G is a single-line IP phone with fixed feature keys. These keys provide one-touch access to the redial, transfer, conference, and voice mail features, Consistent with other Cisco Unified IP Phones, the Cisco 7902G also supports inline power, which allows the phone to receive power over the LAN. This capability gives the network administrator centralized power control, which translates into greater network availability. The Cisco Unified IP Phone 7902G phone supports Cisco prestandard Power over Ethernet (PoE). IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Ine7905G Features + Common-area phone + G.711 and G.729 codecs + Call-monitoring mode ingle line + XML application protocol + SCCP support + Prestandard Power over Ethernet The Cisco Unified IP Phone7905G provides single-line access and four interactive softkeys, which guide the user through call features and functions via the pixel-based LCD. The graph capability of the display provides a rich user experience by presenting calling information, intuitive access to features, and language localization in future firmware releases. The Cisco Unified IP Phone 7905G phone supports Cisco prestandard PoE. This IP phone is appropriate for a common area that does not need a switch port for a PC connection, such as a lobby © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration7910G+SW Features Common-area phone G.711 and G.729 codecs Prestandard Power over Ethernet Call-monitoring mode Single line 802.1 support 10/100 Ethernet switch port No XML application support SCCP support Four programmable keys The Cisco Unified IP Phone 79 10G+SW is a basic telephone that is used primarily in common- use areas (such as lobbies, break rooms, and hallways) that require only basic features. The Cisco Unified IP Phone 7910G+SW includes a Cisco two-port switch, making it suitable for user applications in which basic phone functionality and an Ethemet device such as a PC are necessary. The Cisco Unified IP Phone 7910G+SW supports Cisco prestandard PoE. The Cisco Unified IP Phone 7910G+SW provides four dedicated feature buttons: Line, Hold, Transfer, and Settings. A cluster of six feature access keys is located above the volume control rocker switch. These access keys support message, conference, forwarding, speed dial, and redial features, 16 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc7911G Features Four programmable keys * Single line + Lighted Hold key Call-monitoring function G.711 and G.729 codecs + SCCP support + 802.10 support + 10/100 Ethernet switch port + Power over Ethernet The Cisco Unified IP Phone 7911G is a basic IP phone with an Ethernet switch port, which a core set of business features, This IP phone is basically a Cisco Unified IP Phone 7906 with a switch port. This easy-to-use, display-based IP phone increases productivity while minimizing user training and delivers network and application convergence. The Cisco Unified IP Phone 7911G supports Cisco prestandard PoE. This IP phone is commonly used for basic users who need both a PC and an IP phone. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-17,7912G Features Four programmable keys Single line Lighted Hold key Call-monitoring function G.711 and G.729 codecs SCCP support 802.1Q support 10/100 Ethernet switch port Prestandard Power over Ethernet The Cisco Unified IP Phone 7912G is a basic IP phone with an Ethernet switch port, which provides a core set of business features, This IP phone is basically a Cisco Unified IP Phone 7905 with a switch port. This easy-to-use, display-based IP phone increases productivity while minimizing user training and delivers network and application convergence. The Cisco Unified IP Phone 7912G supports Cisco prestandard PoE This IP phone is commonly used for basic users who need both a PC and an IP phone. 18 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc7920 Features + 802.11b + Vibrate or ring + Lightweight Extensible Authentication Protocol (LEAP) and Wired Equivalent Privacy (WEP) security + Mobility + Qos + G.711 and G.729 codecs + SCCP support The Cisco Unified Wireless IP Phone 7920 is an easy-to-use IEEE 802.11b wireless IP phone that provides comprehensive voice communications in conjunction with Cisco Unified CallManager Express and Cisco Aironet 1200, 1100, 350, and 340 Series of Wi-Fi (IEEE 802.1 1b) access points. The Cisco Unified Wireless IP Phone 7920 delivers seamless intelligent services such as security, mobility, QoS, and management across an end-to-end Cisco network. Note Itis required that you perform a site survey before the use of this IP phone. (© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-197935 and 7936 Features Conferencing G.711 and G.729 codecs 360-degree coverage Power brick required No XML application SCCP support External microphone connection (7936 only) TES The Cisco IP Conference Station 7935 and 7936 models are IP-based, full-duplex conference stations for use on desktops. You can also use these full-featured, hands-free stations in small and medium-sized conference rooms. In addition to the regular telephony keypad, the Cisco IP Conference Station 7935 and 7936 provide three softkeys and menu navigation keys that guide users through call features and functions. The full-duplex design of the Cisco IP Conference Station 7935 and 7936 offers superior voice quality, eliminating echoes, clipped words, and reverberations, for more natural conversation. These conference stations feature superior sound quality with a digitally tuned speaker and three microphones, allowing conference participants to move around while speaking. Note The Cisco IP Conference Stations 7935 and 7936 work best in small and medium-sized conference rooms, rather than large conference rooms. 20 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc.7940G Features + Up to two line appearances + G.711 and G.729 codecs + 10/100 Ethernet switch port + Prestandard Power over Ethernet + XML application support + SCCP support The Cisco Unified IP Phone 7940G is a second-generation, full-featured IP phone for low- and medium-traffic users who require a minimum of directory numbers. It provides two programmable line or feature buttons and four interactive softkeys, which guide users through call features and functions. The Cisco Unified IP Phone 7940G supports Cisco prestandard PoE. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-217941G and 7941G-GE Features Up to two line appearances G.711 and G.729 codecs High-resolution screen 10/100 Ethernet switch port Power over Ethernet XML application support SCCP support GE supports Gigabit Ethernet ‘The Cisco Unified IP Phone 7941G and 7941G-GE add a high-resolution screen to the Cisco Unified IP Phone 7940G full-featured IP phone, primarily for manager and executive needs. ‘They provide six programmable line or feature buttons and four interactive sofikeys to guide users through call features and functions. The Cisco Unified IP Phone 7941G and 7941G-GE support IEEE 802.3af PoE. The Cisco Unified IP Phone 7941G supports prestandard PoE. 1-22 IP Telephony Express (IPTX) v4.0. ‘© 2006 Cisco Systems, inc.7960G Features + Up to six line appearances + G.711 and G.729 codecs + 10/100 Ethernet switch port + Prestandard Power over Ethernet + XML application support + SCCP support ‘The Cisco Unified IP Phone 7960G is a second-generation, full-featured IP phone, primarily for manager and executive needs. It provides six programmable line or feature buttons and four interactive softkeys to guide users through call features and functions. The Cisco Unified IP Phone 7960G supports Cisco prestandard PoE. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-237961G and 7961G-GE Features Up to six line appearances G.711 and G.729 codecs High-resolution screen 10/100 Ethernet switch port Power over Ethernet XML application support ‘SCCP support GE supports Gigabit Ethernet ‘The Cisco Unified IP Phone 7961G and 7961G-GE add a high-resolution screen to the Cisco Unified IP Phone 7960G full-featured IP phone, primarily for manager and executive needs. It provides six programmable line or feature buttons and four interactive softkeys to guide users through call features and functions. The Cisco Unified IP Phone 7961G and 7961G-GE support IEEE 802.3af PoE. The Cisco Unified IP Phone 7961G supports Cisco prestandard PoE. 224 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.7970G Features Up to eight line appearances G.711 and G.729 codecs Color touchscreen 10/100 Ethernet switch port Power over Ethernet External power required for full screen brightness XML application support SCCP support Stereo jack sockets ‘The Cisco Unified IP Phone 7970G demonstrates the latest technology and advancements in VoIP telephony. It not only addresses the needs of the executive or major decision maker, but it also brings network data and applications to users without PCs. This state-of-the-art IP phone includes a backlit, high-resolution color touchscreen display (320 x 234, 12-bit display with 4096 colors) for easy access to communication information, time-saving applications, and feature usage. It also enables customers and developers to deliver more innovative and productivity-enhancing XML applications to the display. Access to eight telephone lines (or a combination of lines and direct access to telephony features), a high-quality hands-free speakerphone, a built-in headset connection, and both Cisco prestandard PoE and IEEE 802.3af PoE are supported, © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-257971G-GE Features + Up to eight line appearances + G.711 and G.729 codecs + Color touchscreen + Gigabit Ethernet switch port Power over Ethernet External power required for full screen brightness XML application support + SCCP support + Stereo jack sockets The first to provide unconstrained bandwidth to desktop applications, the Cisco Unified IP Phone 7971G-GE delivers the latest technology and advancements in Gigabit Ethernet VoIP telephony. It not only addresses the needs of the executive or major decision maker, but it also brings network data and applications to users quickly with its Gigabit Ethernet port for integration with a PC or desktop server. The features of this state-of-the-art Gigabit Ethernet IP phone are identical to those of the Cisco Unified IP Phone 7970G. The Cisco Unified IP Phone 7971G-GE phone also includes a backlit, high-resolution color touchscreen display (320 x 234, 12-bit display with 4096 colors) for easy access to communication information, time-saving applications, and feature usage. It also helps enable customers and developers to deliver more innovative and productivity-enhancing XML applications to the display. Offering access to eight telephone lines (or a combination of lines and direct access to telephony features), a high-quality, hands-free speakerphone, and a built-in headset connection, the Cisco Unified IP Phone 7971G-GE phone can be powered through IEEE 802.3af PoE or a local power supply. The Cisco Unified IP Phone 7971G-GE does not support Cisco prestandard PoE. '26 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.MAI) dateleleetag tard Has pixel-based screen Has multiple softkey buttons along the bottom Displays status of phone Displays call information Can be used to run third-party or custom XML applications ‘The Cisco Unified IP Phone 7935, 7940G, 7941G, 7941G-GE, 7960G, 7961G, 7961G-GE, 7970G, and 7971G-GE models all have a large, pixel-based LCD. The pixel-based LCD displays features such as date and time, calling party name, calling party number, digits dialed, and feature and line status. The four softkey buttons change based on the current state of the call. This feature allows the buttons to be used more efficiently than if they were statically assigned. These buttons can also be invoked and customized by a third-party or a custom XML-based application. Note For more information on XML applications, go to http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a00800 925a8.htm © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-2Expansion Module 7914 Features + Adds on to 7940G, 7941G, 7960G, or 7961G phone Has 14 line appearances or speed dials Connects to the RS-232 port on the IP phone Chains up to two Has lighted button to convey call state Requires new stand Requires power brick ‘The Cisco Unified IP Phone Expansion Module 7914 extends the capabilities of the Cisco Unified IP Phone 7940G, 7941G, 7941G-GE, 7960G, 7961G, or 7961G-GE with additional buttons and an LCD. This expansion module adds 14 buttons to the existing six buttons of the Cisco Unified IP Phone 7960G, 7961G, 7961G-GE models, increasing the total number of buttons to 20 when you add one Cisco Unified IP Phone Expansion Module 7914 and to 34 when you add two Cisco Unified IP Phone Expansion Module 7914 modules. The large LCD of the Cisco Unified IP Phone Expansion Module 7914 enables users to quickly and easily identify associated buttons. Using the Settings menu of the supported IP phone, you can adjust the contrast of the individual LCDs for the IP phone and the Cisco Unified IP Phone Expansion Module 7914, if necessary. ‘You can program each of the 14 buttons on the Cisco Unified IP Phone Expansion Module 7914 as an extension number or a speed dial key, much like an IP phone. In addition, you can use the silent ring option for shared lines mapped to the Cisco Unified IP Phone Expansion Module 7914, the fast transfer capability, and the busy lamp capability to provide attendant console functionality. The Cisco Unified IP Phone Expansion Module 7914 connects to the RS-232 port on the back of the supported IP phone; a new stand and power brick are required. 28 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.ATA 186 and 188 Features + Analog connectivity 186—two analog ports 188—two analog ports plus 10 Fax or analog phone SCCP required for phone H.323v2 support H.323 required for fax The Cisco ATA 186 and 188 Analog Telephone Adaptors connect regular analog phones and fax machines to IP-based telephony networks. Each of the two voice ports on the Cisco ATA 186 and 188 supports independent telephone numbers, giving you two separate lines. In addition, the internal Ethernet switch allows for a direct connection to a 10BASE-T Ethemet network and a 100BASE-TX Ethernet network via an RJ-45 interface. When you use the Cisco ATA 186 or 188 for analog phone connectivity, you should configure it to use SCCP. However, when you use the ATA 186 or 188 for fax connectivity, it must use H.323 connectivity. The two analog ports of the ATA 186 or 188 must both use the same protocol. As a result, you can use the device as either an analog phone or a fax machine, but not both. Note Analog modem connections are supported only on an FXS port local to a router and are not supported on the Cisco ATA 186 of 188. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-29Cisco Unified CallManager Express can use both H.323 and SCCP to control IP phones, analog, phones, and faxes. 230 IP Telephony Expross (PTX) v4.0 (© 2006 Cises Systems, ne.Supported Protocols and Integration Options This topic describes the supported protocols and integration options of Cisco Unified CallManager Express. Skinny Client Control Protocol * Cisco proprietary protocol Call-control protocol Lightweight protocol Low memory requirements + Low complexity Low CPU requirements Cisco Unified CallManager Express software provides call processing for IP phones using SCCP. SCP is the Cisco proprietary protocol for real-time calls and conferencing over IP. This generalized messaging sct allows Cisco IP phones to coexist in an H.323 environment. Savings in memory size, processor power, and complexity are benefits of SCCP. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-31SCCP Phone Limitatior + QoS, bandwidth, and CAC are not supported within SCCP. + Complex connection paths can cause QoS problems. + IP phones are assumed to be connected locally to the Cisco Unified CallManager Express router. + Remote IP phones may be configured as long as latency, jitter, loss, and bandwidth are adequate. QoS, bandwidth management, and Call Admission Control (CAC) are not supported within the SCCP context on Cisco Unified CallManager Express. Complex connection paths could cause QoS problems. Because of these factors, you must connect all IP phones locally to the Cisco Unified CallManager Express router. Remote IP phones may be configured as long as the connection has enough bandwidth, low latency, low packet loss, and low jitter -32 IP Telephony Express (IPTX) v4.0 (© 2006 Cisco Systems, Inc.H.323 Protocol Support for voice, video, and data Industry standard Complex protocol Higher complexity than SCCP CAC functionality Authentication 1.323 is a specification for transmitting audio, video, and data across an IP network, including the Internet, H.323 is an extension of the ITU-T standard H.320. Tip The ATA must be configured with H.323 when fax machines are connected to the analog ports, © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2:33Examples of Recommended H.323 Connections Cisco Unified CallManager Cluster isco Unified CallManager cisco Unified CallManager ‘This figure shows the use of the H.323 protocol to connect the Cisco Unified CallManager Express routers together and to contro! the analog fax connected to the ATA. 234 IP Telephony Express (PTX) v4.0, (© 2006 Cisco Systems, neH.323 Gatekeeper Cisco Unified CallManager Express can register to an H.323 gatekeeper, ensuring that the WAN is not oversubscribed. H.323 ky Register 1000 2 2000 2095551000 3095552000 Register Extension Number, Gatekeeper Register Extension Number, 164 Number, or Both E-164 Number, o Both (oie aimee # You can configure the Cisco Unified CallManager Express system to register an ephone-dn with an H.323 gatekeeper. In addition, the IP phone can have both an extension number and an E,164 number defined, and you can register one or both of those numbers with the H.323 gatekeeper. You can also use H.323 to allow one Cisco Unified CallManager Express to communicate with another Cisco Unified CallManager Express or with voice gateways. You must use a router separate from Cisco Unified CallManager Express if you are going to configure a gatekeeper. The H.323 gatekeeper can provide the following functions: = CAC overa WAN link to ensure that the WAN link is not oversubscribed = Dial plan administration, which centralizes the dial plan for intersite numbering m= IP-to-IP gateway to provide a network-to-network point for billing and security and for joining two VoIP call legs © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2Emerging standard Vendor-specific in most cases igher complexity than SCCP Authentication Based on other well-known protocols SIP was designed as a multimedia protocol that could take advantage of the architecture and messages found in popular Internet applications. By using a distributed architecture with URLs for naming and ASCII text-based messages, SIP attempts to take advantage of the Internet model and standards for building VoIP networks and applications. In addition to VoIP, videoconferencing and instant messaging use SIP. Asa protocol, SIP defines only how sessions are to be set up and torn down. SIP leverages other Internet Engineering Task Force (IETF) protocols to define other aspects of VoIP and multimedia sessions, such as session definition protocol (SDP) for capabilities exchange, URLs for addressing, Domain Name System (DNS) for service location, and Telephony Routing over IP (TRIP) for call routing. “36 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.lame ra ey Cisco Unified CallManager Cisco Unified CallManager Express eS It is possible to use SIP to connect calls between Cisco Unified CallManager Express systems. (© 2005 Cisco Systems, Ine (i800 Unified CallManager Express Configuration 2.37Cisco Unified CallManager Express Requirements This topic describes Cisco Unified CallManager Express requirements. Cisco Unified CallManager bee leoete Celle uta) cy Feature license Seat license Cisco IOS software platform + Release 12.4 (9)T or greater is recommended. + IP voice feature set must be included. Cisco Unified CallManager Express software and files + GUI files + Firmware Cisco Unified CallManager Express requires a Cisco Unified CallManager Express feature license. This license is based on the number of IP phones that you will deploy. The router itself must have a Cisco IOS release that is Cisco Unified CallManager Express-capable. Each IP phone or ATA port also requires a Cisco Unified CallManager Express seat license, which you can purchase with the IP phone, You also need an account on Cisco.com to download Cisco Unified CallManager Express files, such as phone firmware and GUI files and firmware. 2-38 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Cisco Unified CallManager Express Restrictions This topic describes Cisco Unified CallManager Express restrictions. Cisco Unified CallManager bape el cota) eledt [oy + TAPI v2.1 is not fully supported. + Cisco JTAPI is not supported. + Cisco IP SoftPhone is not supported. + MGCP is not supported. There is a subset of TAPI version 2.1 support in Cisco Unified CallManager Express. Cisco Java TAPI (JTAPI) is not currently supported, which restricts the use of a Cisco IP SoftPhone. Also, the newer IP Softphone, the Cisco IP Communicator, is not currently supported, although future versions may be supported. Currently, only third-party sofphones from IP blue work with Cisco Unified CallManager Express. Cisco Unified CallManager Express does not support Media Gateway Control Protocol (MGCP). © 2006 Cisco Systems, Inc. ‘Cisco Unified CallManager Express Configuration 2-39TAPI Lite Functionality Supported: + Operation of multiple independent clients (for example, one client per phone line) + Windows Phone Dialer + Outlook Contact Dialer + Third-party applications Not supported: + TAPI-based softphone + Multiple-user or multiple-call handling (required for ACD) + Direct media and voice handling Cisco Unified CallManager Express does not support TAPI v2.1—Cisco Unified CallManager Express TAPI implements only a small subset of TAPI functionality. It does support operation of multiple independent clients (for example, one client per phone line) but does not fully support multiple-user or multiple-call handling, which is required for complex features such as automatic call distribution (ACD). Applications such as Windows Phone Dialer and Outlook Contact Dialer can use TAPI Lite to dial, place on hold, transfer, and terminate a call on an associated line on an IP phone. JTAPI is not supported, nor are TAPI-based sofiphones. TAPI Lite allows for the control of a line on an associated PC but not for the termination of voice on the PC. Note You can deploy third-party applications to control a line that takes advantage of TAP! Lite. 2-40 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IncSummary This topic summarizes the key points that were discussed in this lesson, BST uir-lp a Cisco Unified CallManager Express is an integrated call- processing solution that is based on Cisco midrange access routers using an optional feature Cisco IOS software. Cisco Unified CallManager Express supports these Cisco platforms: IAD 2430 Series; 1751-V and 1760 Modular Access Routers; 2600 Series Multiservice Routers; 3700 Series Multiservice Access Routers; and 2800 and 3800 Series Integrated Services Routers. Cisco Unified CallManager Express supports all Cisco Unified IP Phones, Cisco Unified CallManager Express software provides call processing for IP phones using SCCP. Certain functionalities are not currently supported in the Cisco Unified CallManager Express software. eee Es (© 2006 Cisco Systems, Ine. ‘Ceca Unified Gallvanager Express Configuration 2-84Lesson 2 Configuring Cisco Unified CallManager Express Network Parameters Overview This lesson describes the Cisco Unified CallManager Express network parameters and the steps to configure these parameters, Objectives Upon completing this lesson, you will be able to configure Cisco Unified CallManager Express network parameters. You also will be able to discuss the need for and the configuration of voice VLANs, DHCP, DHCP relay, NTP, and transcoding between G.729 and G.71. This ability includes being able to meet these objectives: = Describe voice VLANs = Configure voice VLANs on a Cisco Catalyst switch and an BtherSwitch network module m= Identify DHCP service options m= Define a DHCP relay server = Configure NTP m= Describe and configure transcoding between G.729 and G.711Voice VLANs This topic describes voice VLANs. Voice VLANs Prevents unnecessary IP address renumbering ‘Simplifies QoS configurations Separates voice and data traffic Requires two VLANs: one for data traffic and one for voice traffic + Requires only one drop-down Ethernet for the Cisco Unified CallManager Express IP phone and the PC that is plugged into the phone A Cisco IP phone can act as a three-port switch. Just like a switch, the phone can support trunking between itself and another switch. Thus, more than one VLAN can be supported between the IP phone and the access switch into which it is plugged. The three ports of the IP phone are the port that connects to the 10/100/1000 Ethernet switch, the 10/10/1000 Ethernet port into which a PC can be plugged, and an internal port from which voice traffic originates and terminates. The 10/100/1000 Ethernet port that attaches to a switch supports the 802.1Q trunking protocol. This feature enables two VLANs to arrive at the phone, one for the voice traffic and the other for the PC data traffic. The VLAN that the voice traffic goes across is called the auxiliary VLAN, or the voice VLAN. Note Cisco IP phones do not support Inter-Switch Link (ISL) trunking The benefits of this type of configuration include the followin; = This solution allows IP phones to be deployed onto the network without scalability problems from an addressing perspective. IP subnets usually have more than 50 percent — often more than 80 percent—of their IP addresses allocated. A separate VLAN (separate IP subnet) to carry the voice traffic allows a large number of new devices, such as IP phones, to be introduced into the network without extensive modifications to the IP addressing scheme. = This solution allows the logical separation of data traffic and voice traffic, which have different characteristics. This separation allows the network to individually handle each of these traffic types and apply different quality of service (QoS) policies. 2-44 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.™ Because the data and voice traffic are separated, they also can be monitored and managed separately. This solution allows you to connect two devices to the switch using only one physical port and one Ethernet cable between the wiring closet and the IP phone, the PC location, or both. IP Addressing Deployment Options IP Phone + PC on Same Phone + PC on Same ‘Switch Ports. Recommended ‘Switch Ports 177.68.249.100 Ph $ 171.68.249-100 174.68.249.10% wats Public IP Addresses IP Phone Uses Private Notwork IP Phone + PC on Separate Switch Ports IP Phone + PC on Separate Switch Ports 171.68.249.101 | [171.68.249.100 10414 171.68.249.100 =| ad Public IP Addresses Cisco IP phones require network IP addresses. C IP addressing deployment makes the following recommendations for = Continue to use existing addressing for data devices (PCs, workstations, and so forth). Add IP phones using DHCP as the mechanism for obtaining addresses. . = Use subnets for IP phones if they are available in the existing address space. . Use private addressing such as the 10.0.0.0 network (see RFC 1918 for details) if subnets are not available in the existing address space. LANs and private IP WANs will carry these routes between both of the address spaces. The WAN gateway to the Intemet should block private addresses that are currently blocked by data devices. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-4Configuring Voice VLANs This topic describes how to configure voice VLANs on the Cisco Catalyst switch and an EtherSwitch network module. Voice VLANs An access port can handle two VLANs: + Native VLAN + Auxiliary, or voice, VLAN Tagged 802.1@ (Voice VLAN) Soy— Untagged 802.3 (Native VLAN) All data devices typically reside on data VLANs in the traditional switched scenario. You may need a separate VLAN when you combine the voice network with the data network. For configuration purposes, the Cisco Catalyst software command-line interface (CLI) refers to this, new VLAN as the voice VLAN. You can use the new voice VLAN to house nondata devices, in this case, IP phones. The phones will reside in the voice VLAN if you configure the switch to support them; data devices reside in the native VLAN (also referred to as the default VLAN) of the switch. With IP phones residing in a separate VLAN—a voice VLAN—it is easier for you to automate the process of deploying IP phones. The IP phone communicates with the switch via Cisco Discovery Protocol when it powers up. The switch provides the phone with the appropriate VLAN ID, known as the voice VLAN ID (VVID). The VVID is analogous to the data VLAN ID, known as the port VLAN ID (PVID). 2-46 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IncExample Catalyst Switch or EtherSwitch Network Module ‘Console (config) #interface FastEthernet0/1 Console (config-if) #switchport access vlan 12 Console (config-if) #switchport mode access Console (config-if) #switchport voice vlan 112 Console (config-if) #spanning-tree portfast The access VLAN is used for the PC that is plugged into the IP phone. The voice VLAN is used for voice and signaling that originates and terminates on the IP phone. ‘Spanning-tree PortFast mode is automatically enabled when the command switchport voice vian is used. Voice VLANs should be configured on switch access ports. Trunk links do not support voice VLANs even though the configuration is allowed. Use the switchport mode access command to statically configure a switch port in access mode. The command switehport voice vlan vlan-id identifies the VLAN that the switch will provide to the phone using Cisco Discovery Protocol. This command instructs the phone to forward all voice traffic through the specified VLAN and to set a default 802.1Q priority of 5. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-43,Verifying Voice VLAN Configuration Switen chow interface fal/i7 ewiechpore Manes Fa0/17 Switehport: Bnabled Adinistrative mode: static ac Operational Mode: static accel Aaninistrative Trunking Encapmulation: negotiate Operational Trunking Encapsulation: native Access Mode VLAN: 12 (VEANODI2) ‘Trunking Native Mode VLAN: 1 (4 Vosee Vint 122 (ViaNOII2) ‘Trunking VLANe Enabled: ALL Pruning viANs Enabled: 2-1001, fault) Be You can verify your voice VLAN configuration on the Cisco Catalyst switch by using the show interface mod/port switehport command. 248 IP Telephony Express (PTX) v4.0 {© 2008 Gisco Systems, IncRouter Configuration 802.1Q Trunk Trunk on a Router Tarerface fastethornet 1/012 encapeulation dotig 12 Sp address 10.12.0.2 255.255.255.0 interface fastethernet 1/0112 encapsulation dotig 112 Sp address 10.112.0.1 255.255.255.0 Routing between different VLANs requires a Layer 3 router. This router must have an interface that is local to all of the VLANs for which it will route. The most efficient way to get multiple VLANs to the router is to connect a trunk between the switch and the router. This configuration is known as “router on a stick.” The router will have one subinterface local to each VLAN, and only one VLAN can be assigned per subinterface. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-4DHCP Service Setup This topic identifies the DHCP service options. Dynamic Host Configuration Protocol + Assigns IP addresses and subnet masks for one or more subnets. Assigns a default gateway Assigns DNS servers (optional) Assigns other commonly used servers (optional) ‘Scope must be customized to assign a TFTP server to the voice VLAN that IP phones are on Best practice is to configure a DHCP scope for the IP phones DHCP is a very common protocol and familiar to many network administrators. With DHCP, a scope is defined per subnet and is used to assign IP addresses, along with a subnet mask, from a pool of available addresses. If desired, you can assign other values, such as the default gateway and Domain Name System (DNS), to the scope by setting option values. The default gateway option is 003, and the DNS option is 006. ‘These option values can include values specific to an implementation and can be customized by the administrator. Cisco IP phones look for option 150 from their DHCP server. This option contains the IP address of the TFTP server where the IP phone configuration files reside. The administrator must configure option 150 with the IP address of the TFTP server, which is Cisco Unified CallManager Express router. You can deploy DHCP on any platform that supports customized scope options. This includes Windows, Linux, Novell, UNIX, and other operating systems. 2-50 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.DHCP Service Options + Single DHCP IP address pool + Separate DHCP IP address pool for each Cisco IP phone + DHCP relay server You can set up DHCP service for IP phones by defining a single DHCP IP address pool, by defining a separate pool for each Cisco IP phone, or by defining a DHCP relay server. Define a single DHCP IP address pool if the Cisco Unified = Single DHCP IP address poo! DHCP server and if you can use a single shared address CallManager Express router is pool for all your DHCP clients, = Separate DHCP IP address pool for each Cisco IP phone: Define a separate pool for each Cisco IP phone if the Cisco Unified CallManager Express router is a DHCP server and you need different settings on non-IP phones on the same subnet. Note You should avoid separate DHCP scopes for individual devices if possible because of the added configuration complexity. = DHCP relay server: Define a DHCP relay server if the Cisco Unified CallManager Express router is not a DHCP server and you want to relay DHCP requests from IP phones toa DHCP server on a different subnet. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-51Phone Bootup ‘ADHCP scope can be configured on the Cisco Unified CallManager Express router. The scope should define the following: + Range of available IP addresses The IP phone powers on. The phone performs a POST. The phone boots. + Subnet mask Trrough Cisco Discovery + Default gateway rrotocol, the IP phone learns what the woiee VLAN fe + Address of the TFTP server +_DNS server(s) ‘The phone initializes the IP stack After an IP phone receives power, the following happens: = Power-on self-test (POST): The phone performs a set of tests to ensure basic functionality. = Bootup: The phone begins the boot process. Voice VLAN discovery: The phone learns which VLAN is the voice VLAN through Cisco Discovery Protocol. — IP stack ini ing: The phone initializes a basic IP stack. 2-52 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc.Phone Bootup (Cont.) ‘The IP phone sends DHCPDISCOVER broadcast requesting an IP address. The DHGP server selects a free IP 4-| address from the pool and sends it, along with the other scope parameters, as a DHCPOFFER. The IP phone initializes, applying the IP configuration to the IP stack. The IP phone requests a configuration file from the TFTP server. Ea mio ‘The phone bootup process continues with the following sequence: = ~DHCPDISCOVER: By default, the IP phone (DHCP client) sends a DHCPDISCOVER request to the 255.255.255.255 broadcast addres = IP address assigned by DHCP server: If this broadcast is heard by a local DHCP server, the server assigns a free IP address, the subnet mask for the scope, the default gateway for the scope, the DNS server (optional) for the scope, and a TFTP server (option 150) for the scope. =) DHCPOFFER: The scope setting is sent to the DHCP client (the IP phone) using the broadcast address 255.255.255.255. = DHCP settings initialized: The IP phone takes the values received from the DHCP response and applies them to the IP stack of the IP phone. = Configuration requested from TFTP server: The IP phone uses the value received in option 150 to attempt to get a configuration file from the TFTP server (the Cisco Unified CallManager Express router is always the TFTP server in Cisco Unified CallManager Express). ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-53Commands for Manual Configuration cMzrouter (config) # ip dhcp excluded-address start-IP end-IP + Sets a range of addresses to be excluded from the configured scopes cMeRouter (config) # ip dhcp pool pool-name + Creates and enters a DHCP configuration mode cMBRouter (dhep-config) # network subnet subnet-mask + Defines the range of addresses that are available for assignment to DHCP clients ‘Commands for manual configuration are not needed for IP phones if automated setup is used. This is because the setup prompts for these settings and configures a DHCP scope automatically. If'a DHCP scope is not configured or if the administrator wishes to manually configure or change the settings, then these commands must be used. The ip dhep excluded-address start-IP end-IP command allows the administrator to exclude static addresses within the scope range that might be statically assigned to a server or router interface. For Cisco Unified CallManager Express, the exclusions should include the IP address of the router’s interface that may be local to the IP phones. The ip dhep pool pool-name command defines and creates a DHCP pool. After this command has been executed, the router enters a DHCP configuration mode. The automated setup mode creates a DHCP pool named “ITS” (from “Cisco IOS Telephony Service.” as Cisco Unified CallManager Express was formerly known). Note The pool name is case-sensitive Within the DHCP configuration mode under a pool, enter the network subnet subnet-mask command to assign a range of IP addresses to be available for assignment to DHCP clients. This range will not include any exclusion previously defined. When the addresses are assigned, the lowest available IP address is used first. In Cisco Unified CallManager Express, the IP phones are on this subnet. 54 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc.Commands for Manual Configuration (Cont.) cMBRouter (dhep-config) # default-router IP-address + Sets the default gateway that is handed out to the DCHP clients cMzRouter (dhep-config) # dns-server primary-IP [secondary IP] ~ Sets the DNS server or servers that are assigned to the DHCP clients (optional) cMeRouter (dhep-config) # option option-number ip IP-address + Defines a custom option and its value ‘The command default-router /P-address sets option 003 on the DHCP scope that is being defined. This option sends the IP address of the default gateway to the DHCP client. The default gateway for Cisco Unified CallManager Express is the router interface that is on the same subnet as the IP phones. The optional command dns-server primary-IP [secondary-IP] allows the DNS server to be sent in option 006 to the DHCP clients. For Cisco Unified CallManager Express, this setting becomes important if names are used for any of the URL values that can be assigned. Lack of DNS server requires use of IP addresses only Finally, a critical command is option option-number ip IP-address. This is the custom option for the TFTP server. It is important that you configure this command correctly: option /50 ip CallManagerExpress-IP. This IP address must be the IP address on the Cisco Unified CallManager Express router with which the IP phones re © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2:Configuring DHCP on a Cisco IOS Router GiiRouter (contig) #ip dhep excluded-adaress 10.112.0-i 10-112-0.10 cueRouter (config) #ip dhep pool mypoel cuRouter (dhep-config) Wnetwork 10.112.0.0 255.255.255.0 cumRouter (dhep-config) foption 150 ip 10.112.0.1 cuRouter (Ahep-config) Hdefault-router 10.112.0-1 currouter (4nep-config) Hdne-server 10.100.0.1 10.100.0.2 curouter (ahep-config) Hexit + Option 150 sets the TFTP server on the IP phone. + The TFTP server contains the configuration files and firmware for the IP phone. This example shows how to configure the DHCP server with a scope defined for the IP phones. It shows the command option 150 ip 10.112.0.1, where 10.112.0.1 is always set to the IP address of a local interface on the Cisco Unified CallManager Express router that is listening for the TFTP protocol. 2-56 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.DHCP Relay Server This topic defines a DHCP relay server. DHCP Relay Service Cisco Unified CallManager Express Router Without DHCP & oo DHCP Broadcast The router's default behavior is to not forward broadcasts; the DHCP request times out. This issue can be addressed with a DHCP relay server. ‘You must implement a DHCP relay server when the DHCP server does not have a local interface on the network with the DHCP clients. This requirement stems from the broadcast nature of the DHCP request-and-response process. By default, broadcasts do not travel from. one subnet on a router to another subnet on a router. This behavior is a basic characteristic of a router, and changing this behavior effectively turns the router into a software bridge. The way around this issue is to convert certain types of broadcasts to either a unicast or a directed broadcast. This change allows the selected type of broadcast to traverse several routers to reach the destination server or subnet. This is the purpose of a DHCP relay server. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-5DHCP Relay Service (Cont.) Enable DHCP relay on the interface that will receive the DHCP broadcast. DHCP Server a Unicast or Directed DHCP Broadcast Broadcast _— ‘The router forwards. the DHCP request to the DHCP server. The DHCP broadcast request is forwarded through either a unicast or a directed broadcast to the DHCP server. et When the Cisco Unified CallManager Express router is not the DHCP server for the IP phones, there is a good chance that the DHCP server is not local to the IP phones. In this case, the Cisco Unified CallManager Express router—or another device—must convert the DHCP broadcast to a unicast or a directed broadcast. The DHCP request must also be modified to include the originating subnet so that the appropriate scope is selected. When you enable the DHCP relay server on a Cisco IOS router, you enable it on the interface that will be receiving the broadeast. This may or may not be the Cisco Unified CallManager Express router. 58 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc.DHCP Relay Service Feature cuerouter (config) # service dhep + Enables the DHCP server feature on the router (enabled by default) caprouter (config-if)# ip helper-address ip-address + Enables forwarding of select broadcasts to the specified subnet or host The Cisco IOS DHCP server is enabled by default. If it has previously been disabled, use the command service dhep to enable this feature. The command that enables the selective forwarding of certain types of broadcasts is ip helper-address ip-address. You must enter this command on the router interfaces that have IP phones local to them. Example of DHCP Relay Service Enable DHCP relay on the will hear the DHCP broadcast. o]-@ 3 WAN CMERouter (config) #service dhop cMgRouter (config) #interface fastethernet 0/0 cMzRouter (config-if)#ip helper-address 10.200.0.1 This figure shows how to configure an IP helper address of 10.200.0.1 on the FastEthernet 0/0 (£20/0) interface, which is local to the IP phone. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express ConfigurationNetwork Time Protocol This topic describes how to configure Network Time Protocol (NTP). ime Protocol The IP phone gets its displayed time from the Cisco Unified CallManager Express router. The time of the internal clock of the Cisco Unified CallManager Express router should be synchronized with an NTP server. The local NTP server can have an attached atomic clock or can synchronize with a more authoritative source. There are free NTP servers available on the Internet. The time of the Cisco Unified CallManager Express router can be used to stamp all syslog and trace messages. The internal clock of a Cisco IOS router can drift, and a more authoritative source through NTP is desirable. RFC 1305 defines NTP. The heart of the time service is the system clock. The system clock begins to run the moment the system starts. It keeps track of the current date and time. The system clock can be set from a number of sources. It can also distribute the current time through various mechanisms to other systems. Some routers contain a battery-powered calendar system that tracks the date and time across system restarts and power outages. This calendar system is always used to initialize the system clock when the system is restarted It can also be considered an authoritative source of time and redistributed through NTP if no other source is available. NTP, when it is running, can periodically update the calendar which compensates for the inherent drift in the calendar time. When a router with a system calendar initializes, the system clock is set based on the time in its internal battery-powered calendar. On models without a calendar, the system clock is set to a predetermined time constant. NTP allows you to synchronize your Cisco Unified CallManager Express router to a single clock on the network, known as the clock master. Although NTP is disabled on all interfaces by default, it is essential to Cisco Unified CallManager Express. NTP is designed to synchronize the time on a network of machines. NTP runs over the User Datagram Protocol (UDP) which runs over IP. It uses port 123 as both the source and destination. NTP version 3 (RFC 1305) is used to synchronize timekeeping among a set of distributed time servers and clients. An NTP network usually gets its time from an authoritative time source. This can be a radio clock or an atomic clock attached to a time server. NTP then distributes this time across the network. An NTP client makes a transaction with its server over its polling interval which ranges from 64 to 1024 seconds. This interval dynamically changes over time depending on the network conditions between the NTP server and the client. No more than one NTP transaction per minute is needed to synchronize two machines. 2-60 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.NTP uses the concept of a stratum to describe how many NTP hops a machine is from an authoritative time source. For example, a stratum | time server, with a radio or atomic clock directly attached to it, will send its time to a stratum 2 time server through NTP. A machine that runs NTP will automatically choose the NTP configured machine that has the lowest stratum. number as its time source. This strategy effectively builds a self-organizing tree of NTP speakers. (© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-61Europe Configuring the Time cMERouter (config) # clock timezone zone hours-offset + Sets the local time zone cMBRouter (config) # clock summer-time zone recurring [start-date end-date] + Specifies daylight saving time cupRouter (config) # ntp server ip-address + Allows the clock on this router to be synchronized with the specified NTP server The command clock timezone zone hours-offset sets the time zone and number of hours that the time zone is offset from Coordinated Universal Time (UTC) (formerly Greenwich Mean ‘Time [GMT)). This command allows the Cisco Unified CallManager Express router to have its time zone defined. If daylight saving time occurs in the area where the Cisco Unified CallManager Express system is located, then it must be set up using the clock summer-time zone recurring [start-date end-date] command. ‘The command to allow the Cisco Unified CallManager Express router to synchronize with an NTP server is ntp server ip-address. This command allows the Cisco Unified CallManager Express router to keep the correct time based on the time of a more authoritative source than its ‘own system time, The following list of common time zones and their offsets from GMT will help you configure the clock commands. These are the time zones and offsets from GMT for Europe: = GMT Greenwich Mean Time, as UTC = BST British Summer Time, as UTC + | hour = IST Irish Summer Time, as UTC + | hour =) WET Western Europe Time, as UTC m= WEST Western Europe Summer Time, as UTC + | hour = CET Central Europe Time, as UTC + 1 = CEST Central Europe Summer Time, as UTC +2 = EET Eastern Europe Time, as UTC + 2 = EEST Eastern Europe Summer Time, as UTC +3 2-62 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IncUnited States and Canada ‘These are the time Australia These are the time MSK MSD AST ADT ET EST EDT cT csT cDT MT MST MDT PT. PST PDT AKST AKDT HST WST CST EST Moscow Time, as UTC + 3 Moscow Summer Time, as UTC + 4 zones and offsets from GMT for the United States and Canada: Atlantic Standard Time, as UTC ~ 4 hours Atlantic Daylight Time, as UTC ~ 3 hours Eastern Time, either as EST or EDT, depending on place and time of year Eastern Standard Time, as UTC ~ 5 hours Eastern Daylight Time, as UTC — 4 hours Central Time, either as CST or CDT, depending on place and time of year Central Standard Time, as UTC ~ 6 hours Central Daylight Time, as UTC 5 hours Mountain Time, either as MST or MDT, depending on place and time of year Mountain Standard Time, as UTC - 7 hours Mountain Daylight Time, as UTC ~ 6 hours Pacific Time, Pacific Standard Time, as UTC — 8 hours Pacific Daylight Time, as UTC - 7 hours either as PST or PDT, depending on place and time of year Alaska Standard Time, as UTC - 9 hours Al Hawaiian Standard Time, as UTC — 10 hours ‘a Daylight Time, as UTC ~ 8 hours zones and offsets from GMT for Australia: Western Standard Time, as UTC + 8 hours Central Standard Time, as UTC + 9.5 hours Eastern Standard/Summer Time, as UTC + 10 hours (+ 11 hours during summer time) For example, the command clock timezone pst -8 would set the time zone to Pacific Standard Time. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-6:Example of Router Set to PST with Daylight Saving Time Enabled NTP Server ~~@o 8s I 10.41.23 —————— TP phone time comes from the Cisco Unified CallManager Express router time synchronizes with the NTP ‘GikRouter (contig) ¥elock timezone pet -® currouter (contig) #elock aummer-tine zone recurring first sunday april 02:00 last sunday october 02:00 curRouter (config) #ntp server 10-1.2.3 © Unified CallManager Express router in the Pacific Standard 1g time turned on. The router is also set to synchronize its system This example shows the Ci Time zone with daylight s time to that of an NTP server. 264 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Transcoding This topic describes how to configure transcoding between the G.711 and G.729 coder- decoders (codecs), Transcoding Transcoding between G.711 and G.729: Requires hardware-based DSP farm Assists Cisco Unified CallManager Express software ad hoc conferencing when one or more parties use G.729 Call transfer and forward to an endpoint where one leg uses G.729 and the other uses G.711 AG.729 call forwarded to voice mail on the Cisco Unity Express module, which supports only the G.711 codec Sends G.711 MOH feed to a caller who is using G.729 Versions of Cisco Unified CallManager Express prior to Release 3.2 supported G.729 compressed voice calls for two-party calls only. Transcoding between G.711 and G.729 codecs requires a hardware-based digital signal processor (DSP) farm. Cisco Unified CallManager Express Release 3.2 and later support transcoding between G.711 and G.729 for the following features: = Ad hoc conferencing: When one or more remote conferencing parties use G.729. = Call transferring and forwarding: When one leg of a VoIP-to-VolP hairpin call uses G.711 and the other leg uses G.729. (A hairpin call is an incoming call that is transferred or forwarded over the same interface from which it arrived.) = Cisco Unity Express: When an H.323 or session initiation protocol (SIP) call using G.729 is forwarded to Cisco Unity Express. Note that Cisco Unity Express supports only G.711 = Music on hold (MOH): When the IP phone receiving MOH is part of a system that uses G.729 (G.711 MOH is translated to G.729). Because of compression, the MOH that is sent using G.729 loses the fidelity that the MOH has with G.711 ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-6Transcoding (Cont.) DSP hardware for transcoding: NM-HDV (TI-549 DSP) NM-HDV2 (TI-5510 DSP) NM-HD-1V (TI-5510 DSP) NM-HD-2V (TI-5510 DSP) NM-HD-2VE (TI-5510 DSP) PVDM2 slots on the 2800 and 3800 (TI-5510 DSP) DSP chips perform transcoding. The DSP chips are contained on SIMMs or on packet voice/data modules (PVDMs). The SIMMs or PVDMs are then seated in the appropriate slots on a network module or in an onboard PVDM slot like those present on the Cisco 2800 Series routers and the Cisco 3800 Series routers Note Deploying both the TI-549 DSP and the TI-5510 DSP in the same chassis is not recommended. 2-66 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IncTranscoding (Cont.) You can use the DSP calculator that is available at htip://www.cisco.com/publie/support/tac/tools_tab_tools.html to calculate the number of calls that a specific hardware configuration can process. Configuring the NM-HDV Overview: + Configure the location and settings of the voice card. + Configure SCCP parameters on the host router. + Enable the DSP farm and set size. The configuration of the High-Density Voice Network Module (NM-HDV)-based DSP farm ii different from the other DSP farms used by Cisco Unified CallManager Express. The NM- HDV requires that you configure the physical location of the DSP resource and the Skinny Client Control Protocol (SCCP) and that you enable and set maximums of the DSP farm. © 2006 Cisco Systems, Inc. ‘Cisco Unified CallManager Express Configuration 2+Configuring the NM-HDV (Cont.) DsPFarm(config) # voice-card slot + Identifies the slot where the DSP farm is located DsPFarm(config-voicecard) # ‘dsp services dspfarm + Enables the DSP farm services ‘The NM-HDV can be used in the Cisco 2600XM, 2800, 3700, and 3800 Series platforms as a conferencing resource and a transcoding resource. The NM-HDV as a DSP resource is based 0 the TI-529 chip. This section shows the commands that are required to configure the use of DSP resources in Cisco Unified CallManager Express version 3.2 or greater. The first step to configure the NM-HDV as a DSP farm is to use the voice-eard s/o command to identify the slot where the DSP farm resides. This command will also open the voice port configuration mode. When you are in voice port configuration mode, you must enter the command dsp services dspfarm to allow the resource to be used as a DSP farm, 2-68 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IntConfiguring the NM-HDV (Cont.) DsPFarm(config) # scep local interface-type interface-number + Sets the local interface that the transcoding application should use to register with Cisco Unified CallManager Express DsPFarm (config) # seep com ip-address priority priority + Specifies the address and priority where the DSP farm will register DsPFarm(config) # sccp + Enables SCCP and the associated processes Next, use the seep local inverface-type interface-number command to select the interface that the DSP farm will use to register with the Cisco Unified CallManager Express system. The seep cem ip-address priority priority command defines the address of the Cisco Unified CallManager Express system on the DSP farm so that it knows where to register. Set the priority to 1 because there will be only one Cisco Unified CallManager Express router. This makes it the preferred router. Enter the seep command to enable the SCCP processes on the DSP farm router. Note The term “com” as seen in the sccp ccm command usually refers to Cisco Unified CallManager. However, in this case, the command seep ccm should point to the Cisco Unified CallManager Express router because itis the call control device, © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-69g the NM-HDV (Cont.) DsPFarm(config) # dspfarm transcoder maximum sessions number + Specifies the maximum number of sessions supported by the DSP farm DsPFarm(config)# dspfarm + Enables the DSP farm The dspfarm transcoder maximum sessions mumber command specifies the maximum number of transcoding sessions that the DSP farm will support. This number will depend on the number and type of DSP resources present. Finally, use the dspfarm command to configure the NM-HDV as a DSP resource to be used for transcoding and to enable the DSP farm processes on the router. 70 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IncExample of Configuring an NM-HDV-Based Remote DSP Farm emt. Capable Only NM-HDV G71 pgp — Farm DSPFarm (config) #voice-card 1 DsPFarm(config-voicecard) #dsp services dspfarm DsPFarm(config)#sccp local fastethernet 0/0 psPFarm (config) #cep pspFarm(config) #acep ccm 10-1-1.1 priority 1 psPFarm(config) #dspfarm transcoder maximum sessions 12 psprarm (config) #dspfarm In this example, an NM-HDV is installed in a router that is not the Cisco Unified CallManager Express router. The DSP resources are configured to be available for use in transcoding. A device has been configured to use only the G.729 codec because it is located across a low- bandwidth WAN link. This device calls a device that can use only the G.711 codec. The DSP farm provides the transcoding under the direction of the Cisco Unified CallManager Express system. Note Cisco Unity Express supports only the G.711 codec. This is the most common reason for needing the transcoding DSP resources when using Cisco Unified CallManager Express. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-7,Example of Configuring an NM-HDV-Based Local DSP Farm G7t- NM-HDV Capable Only a WAN Pei 10AAA - 6.729 ——— ‘CBRouter (config) #voice-card 1 cMgRouter (config-voicecard) #asp services dspfarm cMgRouter (config) #sccp local fastethernet 0/0 caRouter (config) #scep cagRouter (config) #sccp com 10-1-1.1 priority 1 cMgRouter (config) #dspfarm transcoder maximum sessions 12 cMgRouter (config) #dspfarm In this example, an NM-HDV is installed in the same chassis as the Cisco Unified CallManager Express router. The DSP resources are configured to be available for use in transcoding. A device has been configured to use only the G.729 codec because it is located across a low- bandwidth WAN link. This device calls a device that can use only the G.711 codec. The DSP farm provides the transcoding under the direction of the Cisco Unified CallManager Express system. 1-72 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IncConfiguring the NM-HD-xV, NM-HDV2, and PVDM2 Slots Overview: + Configure the location and settings of the voice card. + Configure SCCP parameters on the host router. + Enable the DSP farm and set size. + Define a DSP farm profile. + Define a Cisco Unified CallManager Express group. Setting up TI-5510-based DSP farms using the One-Slot IP Communications Voice/Fax Network Module (NM-HD-1V), Two-Slot IP Communications Voice/Fax Network Module (NM-HD-2V), and IP Communications High-Density Digital Voice/Fax Network Module (NM-HDV2) involves enabling the DSP farms and SCCP on routers. This configuration includes using the voice-card slot command to define the DSP farm location, using the dsp services dspfarm command to start the appropriate services on the router, and using the seep local interface-type interface-number command to define the local interface to use. Use the seep command to start the SCCP processes. These commands are the same as those used for configuring the NM-HDV and were covered in detail earlier in this lesson. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2Configuring the NM-HD-xV, NM-HDV2, and PVDM2 Slots (Cont.) DsPrarm(config) # dspfarm profile profile-identifier transcode + Enables a DSP farm profile for transcoding DsPFarm(config-depfarm-profile) # codec codec-type + Specifies the codecs supported by the DSP farm The DSP farm profile declares codec usage and the maximum number of transcoding sessions and associates SCCP with the DSP farm profile. This profile is then associated with a Cisco Unified CallManager Express group. The dspfarm profile profile-identifier transcode command creates a profile and enters DSP farm profile configuration submode. Then use the codee codec-type command to define the supported codecs. Note Cisco Unified CallManager Express is capable of controlling transcoding between the G.729 and G.711 codecs only. -74 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IncConfiguring the NM-HD-xV, NM-HDV2, and PVDMN2 Slots (Cont.) DSPFarm(config-dspfarm-profile) # maximum sessions number + Specifies the maximum number of sessions supported by the DSP farm DsPFarm(config-dspfarm-profile) # associate application secp + Associates SCCP to the DSP farm profile While you are in the DSP farm profile configuration submode, use the maximum sessions number command to set the maximum number of simultancous transcoding sessions that the DSP farm allows. Finally, use the associate application seep command to associate SCCP with the DSP farm. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-75Configuring the NM-HD-xV, NM-HDV2, and PVDN2 Slots (Cont.) DsPFarm(config)# scep com ip-address identifier identifier-number + Specifies the IP address of the Cisco Unified CallManager Express router and an identifying number DSPFarm(config) # Scep com group group-number + Creates a Cisco Unified CallManager Express group DSPFarm(config-sccp-ccm) # associate com identifier-number priority 1 + Associates a Cisco Unified CallManager Express router with a Cisco Unified CallManager Express group Only one Cisco Unified CallManager Express group is required. Under the Cisco Unified CallManager Express group, assign a priority to an identifier, associate the group with a DSP farm profile, and set the keepalive, switchback, and switchover parameters. The command seep cem ip-address identifier identifier-number specifies the address of the Cisco Unified CallManager Express router and assigns an identifying number. Then use this number in the associate cem identifier-number priority | command to associate a Cisco Unified CallManager Express router to the Cisco Unified CallManager Express group. A Cisco Unified CallManager Express group is a naming device under which data for the DSP farms is declared. Use the seep cem group group-number command to define the Cisco Unified CallManager Express group. Note The priority should always be set to 1 in a Cisco Unified CallManager Express configuration because the DSP farm can be associated to only one Cisco Unified CallManager Express router, 76 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Configuring the NM-HD-xV, NM-HDV2, and PVDN2 Slots (Cont.) DSPFarm(config-sccp-ccm) # associate profile profile-identifier register device-name + Associates a DSP farm profile with a Cisco Unified CallManager Express group and assigns the registered name DsPFarm(config-sccp-cem) # keepalive retries number + Sets the number of keepalive retries Associate a DSP farm profile to a Cisco Unified CallManager Express group with the command associate profile profile-identifier register device-name. If the number of keepali retries should be set to something other than the default of three, use the keepalive retri number command. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express ConfigurationExample of Configuring the NM-HD-xV, NM-HDV2, and PVDM2 slots NM4HD-AV or G71 NMaID2V or 10.1.4.4 Capable Only “Ninsupve tees. S< G71 psp G.729 — Farm OR NM-HD-1V or NM-HD-2V or G711- MHD 2 Capable Only Grit psp 6-729 Farm ‘The figure shows an example of a router with a TI-5510—based DSP resource installed. The DSP resources are configured to be available for use in transcoding. A device, which has been configured to use only the G.729 codec because it is located across a low-bandwidth WAN link, calls a device that can use only the G.711 codec. The DSP farm provides the transcoding. under the direction of the Cisco Unified CallManager Express system. Example of Configuring the NM-HD-xV, NM- HDV2, and PVDM2 slots (Cont.) DapPars config) Weise cara 7 DsPFarm(config-veicecara) Wasp services depfarn DsPFarm(config) #eccp local fastethernet 0/0 DePFarm(config) #acop com 10.1.1.1 identifier 7 DsPFarm(config) Facop DePrarm(config) #depfarm profile 1 transcode DsPFarm(config-dspfarn-profile) #codec g71iulaw DsPrarm(config-dspfarn-profile) #codec g729ar8 DsPrarm(config-depfarm-profiie) ¥maximum sessions 24 psPFarn(config-depfarn-profile) fassociate application scep DsPFarm(config)#accp com group 22 DsPrarn(config-seep-cen) associate com 7 priority 1 DsPFarn(config-sccp-cem)¥assoctate profile 1 register mtp0006271483°8 DsPParn (config-secp-ccm) #heepalive retries 5 ‘78 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IncConfiguring the Cisco Unified CallManager Express Telephony Service to Use a DSP Farm cupRouter (config-telephony-service) # sdspfarm units number + Specifies the maximum number of DSP farms that are allowed cuBRouter (config-telephony-service) # sdspfarm transcode sessions number + Specifies the maximum number of transcode sessions for G.729 allowed by the Cisco Unified CallManager Express router cusrouter (config-telephony-service) # sdspfarm tag number device-name + Permits a DSP farm unit to register to the Cisco Unified CallManager Express router To utilize the configured DSP farm, you must configure the Cisco Unified CallManager Express router in telephony-service mode. The steps are the same regardless of the type of DSP resource that is configured. The maximum number of DSP farms that may register with the Cisco Unified CallManager Express router is set with the command sdspfarm units number. The default setting is 0. The command sdspfarm transcode sessions number sets the maximum number of G.729 sessions that the Cisco Unified CallManager Express router allows. The range of the command is 0 to 128 sessions and it defaults to 0. The command sdspfarm tag number device-name is given to enable the specific DSP farm to register. The number option is a number from I to 5; the device-name option is the name that the DSP farm will r with and is the MAC address of the SCCP client with “mtp” prepended (for example, mtp00061476aef3). © 2006 Cisco Systems, Inc. ‘Cisco Unified CallManager Express Configuration 2-76The figure shows the configuration in telephony-service mode on the Cisco Unified CallManager Express router that is required to enable the DSP farm to register. 2-80 IP Telephony Express (IPTX) v4.0 © 2008 Cisco Systems, IncVerifying That the DSP Farm Is Registered and Running cMeRouter# show scep [statistics | connections] + Displays the SCCP configuration information and current status cMBRoutert show sdspfarm units + Displays the configured and registered DSP farms cMBRoutert show sdspfarm sessions [summary | active] - Displays transcoding sessions There are show commands available to verify that the DSP farms are configured and registered The first command, show scep [statistics | connections), displays the SCCP configuration as well as information about the past usage of the DSP farm. An example output follows: CMERouter#show sccp statistics SCCP Application Service(s) Statistics: Profile ID:1, Service Type:Transcoding TCP packets rx 7, tx 7 Unsupported pkts rx 1, Unrecognized pkts rx 0 Register tx 1, successful 1, rejected 0, failed 0 KeepAlive tx 0, successful 0, failed 0 OpenReceiveChannel rx 2, successful 2, failed 0 CloseReceiveChannel rx 0, successful 0, failed 0 StartMediaTransmission rx 2, successful 2, failed 0 StopMediaTransmission rx 0, successful 0, failed 0 Reset rx 0, successful 0, failed 0 MediaStreamingFailure rx 0 Switchover 0, Switchback 0 © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-8The command show sdspfarm units displays the configured and registered DSP farms. An example output follows: CMERouter#show sdspfarm units mtp-1 Device:MTP00062714e3fb TCP socket: [2] REGISTERED actual_stream:8 max_stream 8 IP:10.1.1.1 11470 MTP YOKO keepalive 1 Supported codec:G711Ulaw G711Alaw c729a G729ab max-mtps:1, max-streams:24, alloc-streams:8, act-stream: The command show sdspfarm sessions shows the transcoding streams. An example output follows CMERoutertishow sdspfarm sessions Stream-ID:1 mtp:1 10.1.1.1 18404 Local:2000 START usage:Ip-Ip codec: G711Ulaw64k duration:20 vad:0 peer Stream-ID:2 Stream-ID:2 mtp:1 10.1.1.1 17502 Local:2000 START usage: Ip-Ip codec:G729AnnexA duration:20 vad:0 peer Stream-ID:1 Stream-ID:3 mtp:1 0.0.0.0 0 Local:0 IDLE usage: codec:G711Ulaw64k duration:20 vad:0 peer Stream-ID:0 Stream-ID:4 mtp:1 0.0.0.0 0 Local:0 IDLE usage: codec:G711Ulaw64k duration:20 vad:0 peer Stream-ID:0 The variation on the previous command using the show sdspfarm sessions summary command displays a more condensed view of all transcoding streams, An example output follow: CMERouter#show sdspfarm sessions summary max-mtps:1, max-streams:24, alloc-streams:24, act-streams:2 ID MTP State CallID confID Usage Codec/Duration 1 2 IDLE ° G711Ulaws4k /20ms 2 2 IDLE ° G711Ulaw64k /20ms 30 2 START 1 3. MoH (DN=3 , G729 /20ms 42 START -1 3 MoH (DN=3, G711Ulawe4k /20ms 82 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.5 2 IDLE -1 0 G711Ulaw64k /20ms 6 2 IDLE 1 o G711Ulaw64k /20ms The command show sdspfarm sessions active displays the active sessions at any one time. An example output follows CMBRouter#show sdspfarm sessions active Stream-ID:1 mtp:1 10.10.10.3 18404 Local:2000 START usage: Ip-Ip codec:G711Ulaw64k duration:20 vad:0 peer Stream-ID:2 Stream-ID:2 mtp:1 10.10.10.3 17502 Local:2000 START usage: Ip-Ip 729AnnexA duration:20 vad code: peer Stream-ID:1 © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-83Summary This topic summarizes the key points that were discussed in this lesson. Voice VLANs are used to separate voice traffic from data traffic. + Voice VLANs are configured on the interfaces of the switch into which the IP phone is plugged. + Asingle DHCP IP address pool is a large shared pool of IP addresses. Defining a separate pool for each Cisco IP phone creates a name for the DHCP server address pool and specifies IP and MAC addresses for each name. + ADHCP relay server is defined if the Cisco Unified CallManager Express router is not a DHCP server and the DHCP server is not on the same subnet as the DHCP clients. + NTP allows you to synchronize your Cisco Uni Express router to a single clock on the network. DSP resources facilitate transcoding between G.729 and G.711 1d CallManager 2-84 IP Telephony Express (PTX) v4.0 (© 2006 Cisco Systems, neLesson 3 Understanding the IP Phone Registration Process Overview This lesson details the process of registering IP phones with the Cisco Unified CallManager Express router and the files that must be downloaded, Objectives Upon completing this lesson, you will be able to describe the process of registering an IP phone with a Cisco Unified CallManager Express router. This ability includes being able to meet these objectives: = Describe IP phone firmware files and XML configuration files = Describe how Cisco Unified CallManager Express identifies IP phones = Describe how IP phones obtain XML configuration files and IP addressesIP Phone Firmware and XML Configuration Files This topic describes IP phone firmware files and Extensible Markup Language (XML) configuration files. Files Critical to the IP Phone + Firmware + XMLDefault.cnf.xml + SEPAAAABBBBCCCC.cnf.xml Certain files are necessary to the proper operation of the IP phone or analog device so that it can register successfully with the Cisco Unified CallManager Express router: = Firmware: The firmware is loaded into memory on the IP phone and will survive a reboot. = XMLDefault.enf.xml: This XML configuration file specifies the proper firmware, address, and port that the new phone needs to register. =) SEPAAAABBBBCCCC.cnf.xml: This XML configuration file is specific to one device. SEPAAAABBBBCCCC represents the MAC address 2-86 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Firmware + Installed in flash memory with the Cisco Unified CallManager Express software or individually, as needed, on a per-phone basis + Served by the TFTP server on the Cisco Unified CallManager Express router + Uses the command tftp-server flash-firmware-file-name + Multiple firmware files for Cisco Unified IP Phone 7911G, 7941G, 7961G, 7970G, and 7971G models Multiple firmware files may be used on the Cisco Unified IP Phone 7940G and 7960G models Alll of the necessary firmware files for IP phones are stored internally in the flash memory of the Cisco Unified CallManager Express router, so an external database or file server is not required. During registration, IP phones use TFTP to download firmware files from the router flash memory. All Cisco Unified CallManager Express configuration and language files are located in the DRAM of the router under system:/its/. To make the firmware files available through a TFTP server, use the command tftp-server flash:/irmware-file-name. The command load firmware-file-name is also required to associate the model of IP phone with the appropriate firmware file. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-87,The table below lists firmware files based on Cisco Unified IP Phone model, including the Cisco ATA 188 and 186 models and the Cisco Unified IP Phone Expansion Module 7914. These files are specific to Cisco Unified CallManager Express Release 4.0. The files that you need will vary, depending on the version of Cisco Unified CallManager Express that is used. Firmware Files Device Type SCCP Firmware Filename SIP Firmware Filename Cisco ATA 186 ‘ATA030100SCCP040211A.zup ‘ATA030200SIP041111A.zup Cisco ATA 188 ‘ATA030100SCCP040211A.zup ‘ATA030200SIP041111A.zup IP Phone 7902 (CP7902080001SCCP051117A.sbin IP Phone 79056 (CP7905080001SCCP051117A.sbin CP7905010300SIP050414A.sbin IP Phone 7905G (CP7905080001SCCP051117A.2up IP Phone 7910 P00405000700.sbn IP Phone 7910 P00405000700.bin IP Phone 7910 P00405000700.bin IP Phone 7911G term11 default loads ovm11.7-2-0-66.sbn jar 1.7-2-0-66.sbn dsp11.1-0-0-73.sbn apps11.1-0-0-72.sbn cnu11.3-0-0-81.sbn 88 IP Telephony Express (IPTX) v4.0 (© 2006 Cisco Systems, IncDevice Type IP Phone 7912G ‘SCCP Firmware Filename CP7912080001SCCP051117A.sbin ‘SIP Firmware Filename CP7912010300SIP050414A.sbin Expansion Module 7914 $00104000100.sbn Wireless IP Phone 7920 ‘omterm_7920.4.0-02-00.bin Conference Station 7935 00503010100.bin Conference Station 7936 cmterm_7936.3-3-5-0.bin IP Phone 79406 P0030702T023.loads P0030702T023.sb2 P0030702T023.sbn P0S3-07-4-00.loads IP Phone 79406. P0030702T023.bin POS3-07-4-00.bin IP Phone 79416 IP Phone 7941G-GE TERM41,7-0-3-0S.loads term61 default. loads. term41.default.!oads CVM41.2-0-2-26.sbn cnu41.2-7-6-26.sbn Jar41,2-9-2-26.sbn IP Phone 79606, P0030702T023.loads P0030702T023.sb2 P0030702T023.sbn P0S3-07-4-00.loads IP Phone 79606 P0030702T023.bin P0S3-07-4-00.bin IP Phone 79616 IP Phone 7961G-GE TERM41,7-0-3-0S.loads termé61 default. loads term41 default loads CVM41.2-0-2-26.sbn cnud41.2-7-6-26.sbn Jara1.2 -26.sbn IP Phone 79706, IP Phone 79716 TERM70.7-0-3-0S.loads ‘TERM70.DEFAULT.loads TERM71.DEFAULT. loads CVM70.2-0-2-26.sbn cnu70.2-7-6-26.sbn Jar70.2-9-2-26.sbn © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-8Device Configuration XML File SEPAAAABBBBCCCC.cnt. xml" *AAAABBBBCCCC = the MAC address tewcePool “aenagercroun> “Etomborprioriy="0"> SShimanager= Sports 2000 “processNodeName>10-150.t (dan 04 200200:00:00} ‘Soaainformaton>PODSOTOZTO23Closdinformation™ ‘Sisorbocsiom The XML file SEPAAAABBBBCCCC.cnf-xml (where A4AABBBBCCCC is the MAC address of the IP phone) contains the IP address, port, firmware, locale, directory URL, and many other pieces of information. Some of this infor ation cannot currently be used in Cisco Unified CallManager Express. This file is generated during the initialization of the Cisco Unified CallManager Express software if the command ereate-enf-files is in the startup-config file. The figure shows a configuration file that contains the IP address and port that represent the interface with which the phone will attempt to register on the Cisco Unified CallManager Express router, The configuration file also defines the language that will be applied to the IP phone in question 90 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Default XML XMLDefault.cnf.xml IP phones and devices that do not find the more specific SEPAAAABBBBCCCC.cnf.xml file ‘SirernetPnonerort>2000 “processNodeName>10.15..1 ‘alianagerGroup> “oadinformaton®, madel="lP Phone T910">P00409020214«iloadiformations> “oadinfrmation!24 models-Addon T914">cloadhformation 24> ‘Soadintormation.mavel=P Phone 7995"> “Slcadlnormaton’ modei="P Prone Tad0->P003030202\¢ sadinormation? madel="P Phone 1960->Po0%03020214choadnformation?> ‘Phone 19%2°> use the file XMLDefault.cnf.xml. IP phones that download this XML file through TFTP learn the IP address and port of the Cisco Unified CallManager Express router. The IP phones Jearn the version of firmware that is required to function properly with Cisco Unified CallManager Express. The Cisco Unified CallManager Express system generates the file when you enter the command ereate-enf in telephony-service mode, ‘© 2006 Cisco Systems, Inc. ‘Cisco Unified CallManager Express Configuration 29IP Phone Information This topic describes how Cisco Unified CallManager Expres identifies IP phones. ted diel eel cols} + The Cisco Unified IP Phone Expansion Module 7914 cannot autoregister. ‘There is. no Clece Unified 1P Phone Expansion + The Cisco Unified IP Phone Expansion _— Module 7914 in the Module 7914 requires the use of the type *MLDefault.cnf.xmi file command, which is entered by the administrator. + Allother valid devices are recognized automatically by the Cisco Unified CallManager Express system. The Cisco Unified IP Phone Expansion Module 7914 cannot automatically register and requires the use of the type command under ephone configuration mode. None of the other valid IP phones and ATA devices in Cisco Unified CallManager Express require the type command; they are automatically recognized by Cisco Unified CallManager Express. For example, this configuration defines the IP phone with phone tag 10 as a Cisco Unified IP Phone 7960G with an attached Cisco Unified IP Phone Expansion Module 7914: Router (config)# ephone 10 Router (config-ephone)# type 7960 addon 1 7914 2-92 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Downloading and Registration This topic describes how an IP phone obtains its XML configuration file and IP address. Phone Bootup: All Cisco Unified IP Phones Except 7970G and 7971G-GE, Inline Power cd Step 1: Switch sends FLP Stop 2: Ph FLp because of a completed circuit Par 3 ‘Stop 3: Power applied --- ‘Step 4: Link detected on ‘switch port Step 5: P phone boots ‘Step 6: Amount of needed power conveyed ‘through Cisco Discovery Protocol from IP hone to switch Cisco Discovery Protocol These are the steps that take place during phone bootup for all Cisco IP phones when Cisco prestandard Power over Ethernet (PoE) is being used: Step 1 Step 2 Step 3 Step 4 Step 5 Step 6 The switch sends a special tone, called a “Fast Link Pulse (FLP).” out the interface. The FLP goes to the powered device, in this case, an IP phone. The powered device has a physical link when there is no power between the pin on which the FLP arrives and a pin that goes back to the switch, This creates a circuit, and the result is that the FLP arrives back at the switch. This situation will never happen when the attached device is not a PoE-capable device, such as a PC. And if the FLP does not go back to the switch, no power is applied. The switch applies power to the line, The link should go up within 5 seconds. ‘The powered device (IP phone) boots. ‘Through Cisco Discovery Protocol, the IP phone tells the switch specifically how much power it needs. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-93,These Phone Bootup: Cisco Unified IP Phones 7970G and 7971G-GE, Standard-Based Po y. —_-e& Step 1: Constantly sends DC power ” > ‘Step 2-25 ohms of resistance Step 3: 25 ohms of resistance dotected --- Step 4: Low power mode initiated (aw Step 5: P phone boots ‘Step 6: Amount of needed power conveyed ‘through Cisco Discover Protocol from IP. ‘Phone to switch Cisco Discovery Protocol Needed Powar are the steps that take place during phone bootup for the Cisco Unified IP Phone 7970G and 7971G-GE models. Power is the standards-based PoE. Step 1 Step 2 Step 3 Step 4 Step 5 Step 6 ‘The switch constantly applies DC power to all ports that may have a powered device attached to them. The powered device is connected and will have a resistance of 25 ohms if it is PoE- compliant. The switch detects that the device is a PoE-capable device. Power is applied to the link in low power mode, which is 6.3 W. ‘The powered device (the IP phone) boots. ‘Through Cisco Discovery Protocol, the IP phone tells the switch specifically how much power it needs. a4 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Phone Bootup: Cisco Unified IP Phones 7970G and 7971G-GE, Standard-Based PoE (Cont.) DHCP Server DHCP Relay ‘Stop 7: The switch uses Cisco Discovery Protocol to send voice VLAN information to the IP phone. Gisco Discovery Protocol dace Disconeny Protec the IP stack and DHCPDISCOVER broadcast message, roadeast 7 ‘Step 9: The DHCP server hears the DHCPDISCOVER mesa selects an IP address from ‘scope, and sends a DHCPOFFER, _ = — DHCPOFEFR . > IP Address, Subnet Mask, Default Gateway, and TFTP Server (Option 150) Step7 Through Cisco Discovery Protocol, the switch informs the IP phone of its voice VLAN (auxiliary VLAN). Step 8 The IP phone initializes the IP stack and sends out a DHCPDISCOVER broadcast requesting an IP address on the voice VLAN scope. Note Itis possible to hardoode the IP address, subnet mask, default gateway, Domain Name System (DNS), and TFTP server on the IP phone and skip the DHCP steps. However, itis. recommended that you use DHCP to minimize the administrative load that is required to hardcode these settings. Step9 The DHCP server hears the broadcast and assigns an IP address from the scope for the voice VLAN subnet, subnet mask, default gateway, DNS (optional), and address of the TFTP server (the Cisco Unified CallManager Express router). The DHCP server sends back all of the settings to the IP phone in the form of a DHCPOFFER message. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-95Phone Bootup: Known IP Phone (MAC 000F.2470.4A32 Cisco Unified CallManager — Express is the TFTP server. ‘Stop 10: The IP phone 18 addressing information that is obtained through DHCP to the IP stack ‘Step 11: The IP phone looks for an alias named SEPAAAABBBBCCCC.ent.xml (where AAAABBBBCCCC Is the 'MAC address), If the allas Is found, the IP phone registers. TETP Request forthe SEPOOOFZ470AA32 cf xml File ‘midefaut7960.enfxm If no SEP XML file is found, go to Step 14. Ee, Step 10 The phone receives the DHCPOFFER and applies the values ob' Step 11 One of the values carried in the DHCPOFFER message is the address of the TFTP server. The IP phone uses this information to make a connection to the TFTP server and attempt to download the file SEPO00F2470AA32.cnf.xml. The filename is actually an alias to a model-specitic file. This file, if found, contains the information that the phone needs to register with Cisco Unified CallManager Express. This information includes the IP address, port, locale, and firmware file that should be loaded on the IP phone. If the phone has the correct firmware, it registers and gets its configuration. If the firmware is not correct, then proceed to the next step. Ifno SEP XML file is found, go to Step 14. Note The extension numbers, speed dials, and other settings are assigned when the IP phone registers. They are not contained in the SEP XML file. -96 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IncPhone Bootup: Out-of-Date IP Phone Firmware (MAC 000F.2470.AA32 Cisco Unified <2 CallManager Express is the TFTP server. Ot] C ‘Step 12: Ifthe current firmware version is diferent from the ‘version specified in the SEPAAAABBBBCCCC.cnf.xmi file, Tiemware is downloaded from the TFTP server. ETP Request for Firmware. Neoded Firmware Fie ‘Stop 13: The IP phone reboots ifthe firmware was updated. Step 12 If the firmware is out of date or different from the version that is specified, the IP phone goes back to the TFTP server and downloads the appropriate firmware. Step 13 The IP phone reboots after the firmware is downloaded © 2006 Cisco Systems, Inc. ‘Cisco Unified CallManager Express Configuration 2-97Phone Bootup: Unknown IP Phone Unknown IP Address with ‘MAC O00F.2470.4A32 Cisco Unified CallManager Express is the TETP server. ‘Step 14: Ino SEP XML files found, the IP phone ‘downloads the XMLDefault.cnf-xml file from the ‘TETP server. cnt |, _TETP Request forthe XMLDefautenfxmi File ‘Step 15: The phone registers to Cisco Unified CallManager Express, but without any assigned extension. No calls can bo placed of received, and a SEP file will be created on the Cisco id CallManager Express router. or ‘Stop 15: If automatic assignment is enabled or the phone has been Configured, then the new IP phone registers to Cisco Unified CallManagor Express and Is given an extension number. Step 14 If no SEP XML file exists for the specific device, the device is considered new. The new IP phone gets a file called “XMLDefault.cnf.xml” from the TFTP server. The XMLDefault.enf.xml file specifies the IP address, port, and firmware file that the new IP phone needs to register. If the new IP phone has the correct firmware, it can register with Cisco Unified CallManager Express. If it does not have the correct firmware, it will download the correct firmware and reboot. Step 15 The phone registers with Cisco Unified CallManager Express using Skinny Client Control Protocol (SCCP) messages. If automatic assignment is enabled, Cisco Unified CallManager Express assigns an extension. If automatic assignment is not enabled, the phone will have no extension and will not be able to place or receive any calls, 98 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IncSummary This topic summarizes the key points that were discussed in this lesson. BT ure + The IP phone requests the firmware, configuration, and language files when it boots. + Except for the Cisco Unified IP Phone Expansion Module 7914, all valid IP phones and ATA devices are automatically recognized by Cisco Unified CallManager Express. The IP phone uses TFTP-DHCP option 150 to download the information needed to register with Cisco Unified CallManager Express. The IP phone uses its MAC address as part of a created filename to download firmware and configurations and uses the obtained IP address to register with the Cisco Unified CallManager Express router. me {© 2006 Cisco Systems, nc (i800 Unified CallManager Express Configuration 2-99Lesson 4 Defining Ephone-dn and Ephone Overview ‘This lesson defines the Ethernet phone directory number (ephone-dn) and Ethemet phone (ephone) and describes the various types of ephone-dns, Objectives Upon completing this lesson, you will be able to describe an ephone-dn and an ephone and explain how to uilize the different types of ephone-dns. This ability includes being able to ect these objectives: '= Deseribe the purpose of an ephone and an ephone-dn = Define the ephone and describe examples ‘= Define the ephone-dn and describe examples = Describe various types of ephone-dnsWhat Are an Ephone and an Ephone-dn? This topic describes the purpose of an ephone and an ephone-dn, Ephone and Ephone-dn Concepts + An ephone and ephone-dn have modular Cisco 10S software construction. + An ephone represents the physical phone, and the quality is limited by license and hardware. An ephone-dn can be associated with one or more ephones. An ephone can have more than one ephone-dn associated with it. + The maximum number of extensions is the same as the maximum number of ephone-dns. The Cisco Unified CallManager Express software was created with modular and flexible configuration in mind. The composition of the ephone and ephone-dn allows for many types of configurations and designs. The ephone represents the configuration and settings of the physical phone. The ephone is associated with a physical device by MAC address. This Layer 2 address is globally unique. The number of supported ephones on a Cisco Unified CallManager Express system depends on the licensed capacity and the router platform. Currently, it can be no more than 240 ephones. Enterprises with more than 240 phones should use Cisco Unified CallManager. ‘An ephone-dn represents the line or channel that carries voice to the ephone. When you configure an ephone, you can tie the ephone-dn to it. The maximum number of extensions that can be supported at any one time is a quantity determined by the licensed capacity and the hardware platform. When considering the required number of ephones and ephone-dns, you must have this information: = ~=Number of simultaneous calls at each IP phone = Quantity of directory numbers that is desired = Quantity of physical IP phones 2-102 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Ephone-dn This topic defines the ephone-dn and describes examples. Ephone-dn Features Directory number (CLI) and [Rinayeneon pombe extension number (GUI) are | srestletneepronen squivalent oucaliatatng Line and voice port are ee equivalent Sphonedn + Sequence number (GUI) or dn- tag (CLI) is unique and is. Extenslons configured on assigned when tho ephone-an | Seaton sarge is created " see ae enor IN1 and DN2 Can have one or more telephone numbers associated with it ephone-dn Can have one or two voice channels ‘ne phone extension ona] AY) DN1 and DN2 Sin ophone-an for + When itis initially configured, it. | Spnone-an that aoe call creates one or more telephony _ | wating, consultative ‘system POTS dial peers snd conferencing ‘An ephone-dn is software that represents a line that connects a voice channel to a phone instrument on which a user can receive and make calls. An ephone-dn has one or more extensions or telephone numbers associated with it. In most cases, an ephone-dn is equivalent to a phone line. There are several types of ephone-dns with different characteristics Each ephone-dn has a unique dn tag that is a sequence number, that identifies it during configuration. During configuration, you assign ephone-dns to line buttons on ephones. Because each ephone-dn represents a virtual voice port in the router, the number of ephone- WON metatsen [Shanes ‘channel Channel 2 ey su2000 120 Baers Channel ‘Ring No Answor Timeout Same Drecton Nunberon EST] ge fous Seenoneae Globally ‘When you use the no huntstop command on the ephone-din, the call will ring on the first ephone-din, It will then go through any hunting defined on the two channels in a dual-tine ephone-din before the system sends the call to the next most-preferred ephone-dn that has a matching destination patter. This behavior continues until the call reaches an ephone-.tlv . SCCP configuration file SEP.cnf.xml . SIP configuration file SIP.cnf.xml |. SCCP default configuration file SEPDefault.cnf.xmls SIP default configuration file SIPDefault.cnf * First file found is used. universal application loader for phone firmware files allows you to add additional phone features across all protocols, including SIP and SCCP. To do this, a hunt algorithm searches for multiple configuration files. The phone will select the protocol of the first ‘matching configuration file that is found, To ensure that Cisco Unified IP phones download the appropriate configuration files for the desired protocol, SCCP or SIP, you must properly configure the SIP phones before connecting and rebooting the phones. The hunt algorithm searches for files in the following order: 1. CTLSEP file for SCCP phones—for example, CTLSEP003094C25D2E.tlv 2. SEP file for SCCP phones—for example, SEP003094C25D2E.cnf-xml 3. SIP file for SIP phones—for example, SIP003094C25D2E.cnf_ 4. Extensible Markup Language (XML) default file for SCCP phones—for example, SEPDefault.cnf.xmls 5. SIP default file for SIP phones—for example, SIPDefault.cnf Cisco Unified IP phone platforms support both SCCP and SIP. When a SIP phone downloads its configuration profile, the phone compares the phone firmware mentioned in the configuration profile with the firmware already installed on the phone. If the firmware version differs from the version that is currently loaded on the phone, the phone contacts the TFTP server to upgrade to the new phone firmware. The phone downloads the new firmware before registering with Cisco Unified CallManager Express. If you would like to support SIP for any newly acquired Cisco IP phones, the new phone must download the SIP phone firmware before the phone can register. The same requirement applies to all previously configured SCCP phones. After downloading the new firmware, the phone goes through the boot sequence again and registers with Cisco Unified CallManager Express. 150 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc.Loading SIP Firmware on an SCCP IP Phone (Cont.) cusRouter (config-register-global) # upgrade casrouter (config-register-global) # + Performs the TFTP server alias binding create profile + Generates the SIP configuration files The upgrade command performs the TFTP server alias binding, which you can verify with the show voice register tftp-bind command, Then use the create profile command to generate the SIP configuration files. Loading SIP Firmware on an SCCP. IP Phone (Cont.) SIPOOOF2470F45F.cnf + Version of firmware + Phone number on line + Speed dials + Phone features + Softkey buttons + Codec This figure shows an example of a SIP configuration file created for an IP phone with a MAC address of 000F.2470.F45F. Note the version of the firmware, number assigned to line 1, speed dial configuration, and call features. Taage_vereiont P083-07-4-00"7 Linel_name: "2003"; speed_linel: "1010"; call hold_ringback: +0") dnd_control: "0"; anonymous cali block: "0" callerid Blocking: "0" enable vad: "1" semi attended transfer: *1"; call_waiting: efwd_uels enf_join_enab: © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express ConfigurationLoading SIP Firmware on an SCCP. IP Phone (Cont.) ‘GusRouter (config) #vor CusRouter (contig-vos ChtRoster (eontsgvos Cuiskoster (contig vos Cukowter (config) #utep~ CiitRowter (config) #voiee register global Cizkoster(config-regiater global) #mode one CuzRoster (coafig-register global] #ecurce-adéross 10.90.0.1 Cuskouter (cont EmeRoster (eon? ObRoster (config regs Cuzouter (contig-regs: CuERowter (config-rogi "7960-7940 POS3-07-4-00 profiles + All SCP configuration for the desired SIP IP phones must be deleted. + SIP configuration files must be created for the upgrade to be performed. Eta ac This slide shows atypical configuration and the commands to perform an upgrade of firmware toa SIP version of POS3-07-4-00, 2.152 IP Telephony Express (PTX) 4.0 {© 2006 Cisco Systems, in.Loading SCCP Firmware on a SIP IP Phone ‘This topic defines the process to load the SCCP IP phone firmware on a SIP IP phone. Loading SCCP Firmware on an SIP Lear} Converting SIP IP phones to SCCP IP phones: + Delete the voice register pool. = Serve the SCCP firmware through TFTP on the router. + Add an ephone to the configuration. + Ensure that telephony service is set up. = Reset the SIP IP phones. To create an Ethernet phone (ephone) entry and generate a new SCCP XML configuration file for upgrading a particular Cisco IP phone in Cisco Uni SCCP. perform the following tas = Delete the voice register pool for the desired phone or phones. = Serve an SCP version of the firmware via the router TFTP service. = Add an ephone with the associated MAC address or addresses to the configuration, ‘= Ensure that the telephony service is configured. = Reset the IP phone or phones, id CallManager Express from SIP to '© 2006 Cis00 Systems, ne (isco Unified CallManager Express Configuration 2183Summary This topic summarizes the key points that were discussed in this lesson. Tica + Cisco Unified CallManager Express 3.4 and later include ‘Support for RFC based SIP endpoints. + The ability to connect SIP endpoints must be enabled. + Global settings that apply to all SIP IP phones are set in voice register global configuration mode. + The voice register directory number sets the directory number and may be applied to one or more IP phones. + The voice register pool defines the physical IP phone. + To upgrade from the default SCCP firmware on an IP phone to SIP requires that new configuration files be written. + To change from SIP to SCCP firmware requires the deletion of the voice register pool and the configuration of the ephone. Sa TAT 2.154 IP Telephony Express (PTX) v4.0 (© 2006 Cisco Systems, neLesson 6 Describing Cisco Unified CallManager Express Files Overview This lesson describes the Cisco Unified CallManager Express files and how to work with them. Objectives Upon completing this lesson, you will be able to describe Cisco Unified CallManager Express ‘methods for downloading files to IP phones. This ability includes being able to meet these objectives: Describe downloading bundled and individual Cisco Unified CallManager Express files Identify Cisco Unified CallManager Express GUI files to enable web access Identify the Cisco Unified CallManager TSP files for TAPI integration Describe the MOH and xml.template filesCisco Unified CallManager Express Files This topic describes Cisco Unified CallManager Express files, Cisco Unified CallManager Express Files TFTP or FTP Server GurFites Firmware Music on Hold Cisco 105, isco 108 copy tftp flash or copy ftp flash + Load firmware for IP phones and devices + Used to upgrade Cisco Unified CallManager Express + Load music on hold files Cisco Unified CallManager Express requires you to copy firmware files to the flash memory on your router and share them using TFTP or FTP. Download Cisco Unified CallManager Express Release 4.0 files to a TFTP or FTP server that is accessible to your Cisco Unified CallManager Express router. To move the files from the server to the flash memory, use the copy tftp flash command or the copy ftp flash command. You can download the files in a single bundle or individually When you upgrade the Cisco Unified CallManager Express router, you must move the new files, such as firmware, GUL files, and Cisco IOS software, to the flash memory on the router. You may need to periodically update other files, such as new firmware versions and music on hold (MOH) files. 2:56 IP Telephony Express (PTX) v4.0 (© 2006 Ciseo Systems, nc.Bundled Cisco Unified CallManager Express Files This subtopic describes downloading bundled Cisco Unified CallManager Express files. Bundled Cisco Unified CallManager Express Files sot by: Filename ¥ Filename : Release | Dae Tees 40 28-JUN-2006 | 22905929 1 |S 12.4G)T releases ‘You can download a bundled file with all of the Cisco Unified CallManager Express files fron http:!/www.cisco.com. The Cisco Unified CallManager Express bundle comes in a zip file. You can then extract these files from the FTP or TFTP server. Tip You can find the Cisco Unified CallManager Express software at http://www. cisco.com/kobayashi/sw-center/sw-voice.shtml, © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-1Bundled Cisco Unified CallManager Express Files (Cont.) fan 7o7b backgrounds [3 araosoi00sccrosoci 1a ap scolbedMETSPSetup201 exe i cne-bacd2.1.0.0tar cme gut 0.0.1.3 icra 794 sep 7.21. 8 enkr-7941-7961-secp, 7.0.3 tar i crter 7970-71-29. 7 03a enter_79004,002-05.80 The extracted | rcrmy336 32-0 cme-124-9T.zip -730208000152CP051117A sbi. “men Fe cpresreonisceosiiitacon, file yields: [3 cpreccooo0isccP0stit7A zw cPotzoon scot 78sn fesmuscon tad ayreoorerae3 tar | Arcowsccoran bn FB reoweon700 tn | Aronensioio0. bn FAecoonerx0 02 nator 8 snscc100 son * All files are specific to the version of Cisco Unified CallManager Express. eee The Cisco Unified CallManager Express bundle contains all of the files that you need to install and configure Cisco Unified CallManager Express. The figure lists the files that are contained in the bundle, The cme-124-9T zip file contains all the files needed to run the GUI web interface for Cisco Unified CallManager Express. The GUI of Cisco Unity Express also needs these files. You can use the music-on-hold.au file to provide MOH from a file in flash memory. You can replace it with a custom .wav or .au file if desired. 158 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IneIndividual Cisco Unified CallManager Express Files This subtopic describes downloading individual Cisco Unified CallManager Express files. Individual Cisco Unified CallManager Express Files ware files Basic bundle GUI bundle + Phone backgrounds Ringtones + Cisco Unified CallManager TSP Basic automatic call distribution (B-ACD) ‘CME 410 basic system Ses fr 10S 12 4997 e-qyed 0 ttt (CME 40-GUI fos for 10S 12.40) eoaser ue 30 11a Tor 0S 12 46)Teleases Samal Phone Ping tones 270 bacqsounds ar ‘Sample Back god Ses fr 727071 Satan nape Express Telephony ‘0 - Compare CME Verson 40, Frrmwave esr 79607980 - Companble OME ‘4001 You can download the files individually as well as in a bundle. si Byes 2a un-zo06 | sex0000 Date 2B.0unt2005 | ene860 May 2006. ensBGe eMaR 2006 aaa 2AFEB.aMe. 622880 S-FEB. ane esis Note These files are specific to Cisco Unified CallManager Express 4.0, and they are not backward-compatible. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2.159GUI Files This topic identifies Cisco Unified CallManager Express GUI files to enable web access. 400 tar 400) ZevuName som (CME 40 basic system les fr 10S 12.4097 ‘inegud O0 ar 40g) | 2eWuN2O06 BIE CHE 40 GUI fies for 10S 12 40 releases EME 34 GUI fis for 10S 12 4 releases ange tr TeMan ane See Samole Phene Rng ores ftv backgounds at 27-FE6-2006 Gris ‘Sample Backgfound les fr 970771 TFFEG-2006 055058 ‘Semee Provider 20- Compatitie CME ‘Version 40,33 72) 27FEB2008 948736 7960/7940 - Compatible CME ‘Version 4 One of the individual files that you can download is the .tar file that contains the GUI web interface for Cisco Unified CallManager Express. The CUE module GUI is also dependent on the Cisco Unified CallManager Express GUI. 2-160 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc.GUI Files (Cont.) The extracted cme-gui-4.0.0.1.tar yields: XML template xmltemplate Gui files, ‘admin_user.html ‘admin_userjs CiscoLoge.git Delete gif domjs downarrow.gif ephone_admin.htm! ormal_userjs Plus.gif + sxiconad.git Tab gif tolephony_service.html uparrow.gif| amitest htm This figure shows the contents of the GUI web interface .tar file. These files need to be presen in the flash memory of the Cisco Unified CallManager Express router. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 24Cisco Unified CallManager Express—TAPI Integration This topic identifies Cisco Unified CallManager Telephony Service Provider (TSP) files for Telephony Application Programming Interface (TAPI) integration. Cisco Unified CallManager Express—TAPI Integration T7FEB200 | 622592 27-FEB.2006 4085038 Semice Pronder 20 - Compatible CME ‘version 40,33 Tad) ar FEB2006 348736 Firmware files for 7960/7940 - Compatible CME Version 4.0 TAPI Lite + Allows third-party software to control an IP telephony device + Is installed on Windows PC To allow third-party software to interact with the Cisco Unified CallManager Express system through TAPI Lite, you must install the files in the Cisco IOS TSP file on the same Windows PC where the software is installed. The figure shows the contents of the Cisco IOS TSP file. Run CiscoUnifiedCMETSPSetup. on the Windows PC where you are performing the TAPI integration. This file is compatible with Cisco Unified CallManager Express Release 3.3 and 4.0. Note This file does not need to reside in flash memory; it will be extracted and installed on a Windows PC. 2462 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems,Additional Files This topic describes the music-on-hold.au and xml.template files. Additional Files music-on-hold.au + Use the music-on-hold.au audio file to provide music for external callers who are on hold when you are not using a live feed. xml.template + Use the xml.template file to allow or restrict the GUI functions that are available to an optional customer administrator. vase Other files that may be of interest include the file needed for MOH. This file must reside in flash memory on the Cisco Unified CallManager Express router and must be named “music-o hold.au.” The file, which came in the bundle or that you downloaded individually, contains an audio file that is used when a caller is placed on hold. You can customize this file: ‘A sample file for creating a customer administrator with a limited subset of administrative privileges is included in the bundle, or you can download it in an individual file that contains the basic files. You can customize this file, xml.template, and store it in flash memory for us © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2Summary This topic summarizes the key points that were discussed in this lesson, Sure * Use the copy command to move files to the flash memory of, the Cisco Unified CallManager Express router. + Files can be downloaded individually or as bundled. + The files may be compressed and may have to be extracted. + Files that are downloaded include the basic files for Cisco Unified CallManager Express, GUI web interface, TAPI integration, MoH, and the xml.template file. Set a ee 2.164 IP Telephony Express (IPT) va.0 (© 2006 Cisco Systems, ine.Lesson 7 Understanding Initial Phone Setup Overview This lesson describes the three ways to create an initial IP phone setup. It also discusses optional parameters, the commands for rebooting IP phones, setup troubleshooting, and the steps for verifying the Cisco Unified CallManager Express phone configuration. Objectives Upon completing this lesson, you will be able to configure initial IP phone setup and verify Cisco Unified CallManager Express configurations, This ability includes being able to meet these objectives: = Describe the three ways to create an IP phone setup in a Cisco Unified CallManager Express system = Perform a manual setup using the router CLI = Perform a partially automated setup using the router CLI = Perform an automated setup using the Cisco Unified CallManager Express setup tool Describe and use QCT © Identify optional IP phone parameters Discuss two ways to reboot IP phones = Describe troubleshooting tips = Describe the steps to verify Cisco Unified CallManager Express configurationSetting Up Phones in a Cisco Unified CallManager Express System This topic describes the four ways to create an initial IP phone setup in a Cisco Unified CallManager Express system. Four Ways to Set Up Phones + Manual Requires numerous commands from the CL! Requires knowledge of Cisco Unified CallManager Express commands ~ Requires that phones be entered manually in Cisco 10S software + Partially automated Requires numerous commands from the CL! Requires knowledge of Cisco Uni ~ Simplifies deployment + Automated Needs few commands from the CLI Requires little knowledge of Cisco Unified CallManager Express commands ~ Simplifies deployment + Quick Configuration Too! Reduces time required to deploy and configure ~ Requires little knowledge of Cisco Unified CallManager Express commands ~ Web-based too! There are four ways to set up IP phones in Cisco Unified CallManager Express. You can set up phones manually. You can use a combination of manual setup and automated setup, referred to as “partially automated.” You can use fully automated setup. You can also use Cisco IP Communications Express Quick Configuration Tool (QCT). 2-166 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc,Manual Phone Setup how to perform a manual phone setup in a Cisco Unified CallManager the router command-line interface (CLI). This topic describes Express system usi Manual Setup Overview All commands can be entered from the CLI. Manual setup is best performed by experienced administrators. + Administrators leverage their knowledge of Cisco 10S software. Full functionality is achieved through Cisco 10S commands. Deployment of IP phones can be batched or scripted through a text file. The manual setup of the Cisco Unified CallManager Express system involves using the CLI This type of setup allows the administrator to leverage existing knowledge of Cisco 10S software and to implement Cisco Unified CallManager Express functions. You ean view, back up, and restore the configuration through a simple text file. Manual setup can save time and effort when used for multiple site deployments because it allows you to change only the differences on a per-site basis. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-16You must configure the following commands to deploy a Ci Commands Overview Commands that are needed to configure a basic telephony service are as follows: {ftp-server flash:filename telephony-service max-ephones max-ephones max-dn max-directory-numbers load phone-type firmware-file ip source-address ip-address [port port] create onf-fles, keepalive seconds dialplan-pattern tag pattern extension-length length extension-pattern pattern system. = tftp-server flash:filename = telephony-service = max-ephones max-ephones = max-dn mar-directory-numbers = load phone-type firmware-file = ip source-address ip-address port port] = create enf-files = keepalive seconds . 0 Unified CallManager Expre plan-pattern fag pattern extension-length /ength extension-pattern pattern In addition to these commands, you must manually configure Ethernet phones (ephones) and Ethernet phone directory numbers (ephone-dns), 168 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inctftp-server Command cMpRouter (config) # tftp-server flash: filename + Allows a file in flash to be downloadable with TFTP Available through TFTP [eftp-server flash+P00303020214.bin tftp-server flash: emterm 7920.3.3-01-06.bin, jtftp-server flash:P00403020214-bin The command tftp-server flash:/i/ename allows the specified file that resides in flash memory to be downloaded via TFTP. In Cisco Unified CallManager Express, you need to configure the firmware files so that they are available through TFTP. The figure shows firmware for the Cisco Unified IP Phone 7910G+SW, 7920, 7940G, and 7960G models. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-169BE) irda acted (ee ere lrc Lele} cMERouter (config) # telephony-service + Enters telephony-service mode cMERouter (config-telephony-service) # max-ephone maximum-ephones + Sets the maximum number of ephones that may be defined in the system (default is 0) césRouter (config-telephony-service) # max-dn maximum-directory-numbers + Sets the maximum number of ephone-dns that may be defined in the system (default is 0) Issue the telephony-service command to enter telephony-service mode, from which you enter much of the configuration for the Cisco Unified CallManager Express system. The first two commands that you should enter are max-ephones and max-dn, Both of these commands are set to 0, which has the effect of not allowing any ephones or ephone-dns to be configured. Set these commands to the required number of phones and directory numbers. Setting this number greater than the required amount will cause the router to reserve memory for these nonexistent devices. The number of ephones and ephone-dns is version- and platform-specific. The number displayed in Cisco IOS software Help is not always accurate and may reflect an artificially high number. Consult the information provided with the Cisco Unified CallManager Express router or on Cisco.com, Example This is an example of Cisco IOS software Help information that may be displaying maximums higher than the platform can handle. CMBRouter (config-telephony) fimax-dn ? <1-288> Maximum directory numbers supported CMERouter (config-telephony) #imax-ephones ? <1-100> Maximum phones to support. 2170 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, inc.Firmware Association cMsRouter (config-telephony-service) # load model firmware-file + Associates a firmware file with the model of IP phone 7940G and 79606 Liman) - load 7960-7940 P00303020214 load 7920 cmterm_7920.4.0-01-08 =] 7920 load 7920 700402020214 — Filenames are case-sensitive. 7910G+SW To associate a type of Cisco IP phone with a phone firmware file, use the load model firmware-file command in telephony-service configuration mode. The following shows the supported phone models for which you can load firmware: Note Do not use a suffix when using the load command for the Cisco Unified IP Phone 7910G+SW, 7940G, and 7960G models, = 7902 Selects the firmware load file for the Cisco Unified IP Phone 7902G model = 7905 Selects the firmware load file for the Cisco Unified IP Phone 7905G model = 7910 Selects the firmware load file for the Cisco Unified IP Phone 7910G+SW model = 7912 Selects the firmware load file for the Cisco Unified IP Phone 7912G model = 7914 Selects the firmware load file for the Cisco Unified IP Phone Expansion Module 7914 = 7920 Selects the firmware load file for the Cisco Unified Wireless IP Phone 7920 model = 7935 Selects the firmware load file for Cisco Unified IP Conference Station 7935 model = 7936 Selects the firmware load file for Cisco Unified IP Conference Station 7936 model = 7960-7940 Selects the firmware load file for the Cisco Unified 7960G and 7940G models = ATA Selects the firmware load file for Cisco ATA 186 and 188 Analog Telephone Adaptors To see a list of phone models supported by your router, enter the following: CMERouterl (config-telephony) #load ? ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-171EeSel cet cacti l agelad cusRouter (conf ig-telephony-service) # ip source-address ip-address [port port] + Identifies the address and port through which IP phones communicate with Cisco Unified CallManager Express _ ee [ephony-service ip source-address 10.90.0.1 port 2000 The Cisco Unified CallManager Express system expects to receive Skinny Client Control Protocol (SCCP) messages from the IP phones concerning registrations and call control. Use the ip source-address ip-address [port port] command to configure the local IP address and the TCP port from which the Cisco Unified CallManager Express system expects these messages, The port by default is set to 2000; although you can change this designation, it is unusual to do so, Example This is an example of the XMLDefault.cnf.xml file. Note the IP address, port, and firmware files: «member priority="0"> 2000 10.90.0.1 P00403020214 2172 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc P00303020214 cmterm_7920.3.3-01- 06 .bin Create XML Files cupRouter (config-telephony-service) # create cnf-files + Builds the specific XML files that are necessary for the IP phones hy ser | SEPOOOF2473AB14.cnt.xmi 10.90.01 ‘telephony-service create cnf-files Use the create enf-files command in telephony-service configuration mode to build the Extensible Markup Language (XML) configuration files that the IP phones require and that are used with Cisco Unified CallManager Express When you enter this command, the file XMLDefault.cnf.xml is generated with the appropriate settings, including the firmware defined by the load command, the IP address that the new IP phones will be registered with, and the TCP port on which the SCCP messages will arrive. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-173Example This is an example of SEPO00F2473AB14.cnf-xml. Note the IP address, port, locale information, and required firmware: 2000 10.90.0.1 {Jan 01 2002 00:00:00} P00303020214 English_United_States en United_States 0 http://10.90.0.1/localdirectory 24174 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IncKeepalive cupRouter (config-telephony-service) # keepalive seconds + Sets the time interval between keepalive messages from the IP phones to Cisco Unified CallManager Express ‘telephony-service keepalive 10 weeps + Default is 30 seconds, range is 10-65,535 seconds + If three successive keepalives are missed, device must register again To set the length of the time interval between successive keepalive messages from the Cisco Unified CallManager Express router to IP phones, use the keepalive command in telephony- service configuration mode. The default setting for the keepalives is 30 seconds. If the router fails to receive three successive keepalive messages, it considers the phone to be out of service until the phone reregisters. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-175Direct Inward Dialing Configuration Commands cMBRouter (config-telephony-service) # @ialplan-pattern tag pattern extension-length length extension-pattern pattern [no-reg] + Sets a dial plan pattern that can expand extension numbers to fully qualified E.164 numbers, which can be used for DID numbers. Extension 1000 PSTN: -SONPRI_) oo" ‘ ID Numbers Assigned. 2015569000 Extension “Through 1099 2013559009 ‘telephony-service Aialplan-pattern 1 20155590.. extension-length 4 extension pattern 10. You enter directory numbers for the Cisco IP phones in extension-number format. The jialplan-pattern command creates a global prefix that you can use to expand the abbreviated extension numbers into fully qualified E.164 numbers. The dial-plan pattern is also required for registering Cisco IP phone lines with a gatekeeper. The dialplan-pattern command can transform an incoming call that has a full E.164 number to a Cisco IP phone extension number. The extension-length keyword enables the system to convert a full E.164 telephone number back into an extension number for the purposes of caller ID display and received-call and missed-call lists. For example, a company uses the extension number range 100 to 199 across several sites, and the extensions from 1000 to 1099 are present only on the local router. An incoming call from 1044 arrives from the internal VoIP H.323 network of the company—the calling number for this call will display as 408555 1044 in its full E.164 format. By default, the numbers matching the dialplan-pattern command will register to an H.323 gatekeeper if'a gatekeeper is configured. Use of the no-reg keyword changes this default behavior and prevents the numbers that match the pattern from registering with the gatekeeper. When the called number matches the dial-plan pattern, the call is considered a local call and has a distinctive ring that identifies the call as internal, Any call that does not match the dial-plan patter is considered an external call and has a ring that is different from the internal ring. The valid dial-plan pattern with the lowest dial-plan tag number is used as a prefix to all local Cisco IP phones ‘The number of extension-pattem characters must match the extension length that you specify in the dialplan-pattern command. Note You can use this command in place of configuring secondary numbers on ephone-dns. “176 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Example: Manual Setup of Cisco Unified CallManager Express See the lesson “Defining Ephone-dn and Ephone’- for manual configuration information. Sab incon Example: Manual Setup of Cisco Unified CallManager Express This figure shows the configuration for a basic Cisco Unified CallManager Express system. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2477Partially Automated Phone Setup This topic describes how to perform a partially automated IP phone setup in a Cisco Unified CallManager Express system using the router CLI Overview of Partially Automated Setup + Ina partially automated setup, you do not have to configure ephones. + Deployment of IP phones is automated. + The auto assign command is used. + All ephone-dns must be the same type {single-line or dual-line). In a partially automated setup, you do not have to configure ephones. The setup can automatically detect the ephones and it can assign an ephone-dn from a range of configured ephone-dns (all ephone-dns must be the same type). The partially automated setup allows for the deployment of many phones without the work of configuring every phone manually. Use the auto assign command to perform this automatic assignment. 2-178 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc.auto assign Command casnouter (conf ig-telephony-service) # auto assign start-dn to stop-dn [type phone-type] [ctw number timeout seconds] The ephone-dns that are configured to new ephones are automatically assigned. Phones can take up to five minutes to register. Wait for all phones to register before saving the configuration. The cfw and timeout keywords define the Call Forward Busy number and timeout values for phones that register. To automatically assign ephone-dn tags to Cisco IP phones as they register for service with the Cisco Unified CallManager Express router, use the auto assign command in telephony- service configuration mode. This command lets you assign ranges of ephone-dn tags according to the physical phone type. You can use multiple auto assign commands to provide discontinuous ranges and to support multiple types of IP phones. You may assign overlapping ephone-dn ranges so that they map to more than one type of phone. If you do not specify a type, the values in the range are assigned to phones of any type, but if you do assign a specific range for a phone type, the available ephone-dns in that range are used first. The efw and timeout keywords set the Call Forward Busy number and timeout values on all phones that automatically register. You cannot use the auto assign command for the Cisco Unified IP Phone Expansion Module 7914, You must manually configure phones with one or more expansion modules. Automatically assigned ephone-dn tags must belong to normal ephone-dns and cannot belong, to paging ephone-dns, intercom ephone-dns, music on hold (MOH) ephone-dns, or Message Waiting Indicator (MW) ephone-dns, The ephone-dn tags that the system automatically assigns must have at least a primary number defined, All the ephone-dns in a single automatic assignment set must be of the same kind (either single-line or dual-line). Automatic assignment cannot create shared lines. If there are not enough available ephone-dns in the automatic assignment set, some phones will not receive ephone-dns. You must manually reverse the automatic assignment using the CLI. You must follow this reversal by a reboot of the phones that are assigned. If you use the type keyword with this command, use the reset command to reboot the phones. If you do not use the type keyword with this command, use the restart command to perform a quick reboot. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-17Note You should take care when using the auto assign command because this command grants telephony service to any IP phone that attempts to register. If you use the auto assign ‘command option, make sure that your network is secure from unauthorized access by unknown IP phones, Example: auto assign Command When a new IP phone registers with a Cisco Unified CallManager Express system, a new ephone is Telephony service created with the MAC address of the IP phone. ‘auto assign 1 co 10 type 7920 An existing ephone-dn is assigned to the new ephone | auto aseiga 11 co 20 type 7940 from the range defined for the type of phone. favto assign 21 to 40 type 7960 ‘The lowest unassigned ephone-dn in the matching auto assign 42 to 50 statement range is used. If all ephone-dins in a range have been assigned, ‘some phones may not receive an ephone-dn or may overflow to the general automatic assignment without atype. Ifa new IP phone does not match any auto ‘a type, the automatic assignment without a type is used. ae as ‘ephone-dn 2 dual-1ine In this example, there are four auto assign commands with a different ephone-dn assigned to each, The system will assign any Cisco Unified Wireless IP Phone 7920 the lowest unassigned ephone-dn from | through 10. The system will assign any Cisco Unified IP Phone 7940G the lowest unassigned ephone-dn from 11 through 20, and the system will assign any Cisco Unified IP Phone 7960G the lowest unassigned ephone-dn from 21 through 40. And finally, the system will assign any Cisco Unified IP Phone 7920, 7940G, and 7960G an ephone- dn from the generic range of 41 through 50 if it cannot be assigned an ephone-dn in its assigned range. The system will also use this generic range, which is not tied to any type, for any other unspecified models of IP phones. Note When all desired IP phones have been automatically assigned, be sure to save the configuration 80 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc.Automated Phone Setup This topic describes how to use the setup utility to perform an automated IP phone setup in a Is simple to configure Has a question-and-answer interface Is designed for inexperienced administrators Creates Cisco 10S commands in the background Automates deployment Must be no existing telephony service configuration een Automated setup is useful for the administrator who does not have much experience with Cisco 10S software and who may not feel comfortable manually configuring the Cisco Unified CallManager Express system. A question-and-answer interface starts the process—the administrator only has to provide appropriate answers to the questions. Note You must remove any existing configuration of the telephony service in Cisco Unified CallManager Express prior to starting the setup. © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-181Running the Automated Setup Utility Configure NTP prior to. Casein nar Calg aaphony Nie running the setup uty. [Senenssateen terry Load the firmware files | Covigunng ONCP Poo! or laco1OS Telephony Serices into flash memory prior to | “network tr ttphony service NCD Poa Ta.900.0 running the setup utility. ‘Subnet mask for DHCP network :255.255.255.0 Enter the automated setup | 1F7P Sever IP ass Option 150) 1090.03 mode using the telephony- | Do you want to start telephony-service setup? [yesino]: y service setup command, | eantgurng cece 0s Telephony Sevices Entorth aource aderee for Caco 08 Telephony Series 10.0004 + Aquestion-and-answer | Enerite Skim Porter sot08 Telephony Sersces (200 2800 session starts, asking for | How many IP phones do you want to configure : [0]: 10 basic parameters. Dovouwan! duane extensions assigned to phone” sino]: y __ | wat Language doyou want on phones The Ctrl-C keystroke pair © English 6 Dutch can be used at any time to | 1 French 7 Norwegian interrupt or exit the setup | 2 German 8 Portuguese 3 Russian 9 Danish utility. 44 Spanish $0 Swedish No changes are 5 italian committed until the end. L!? The Cisco Unified CallManager Express setup utility provides a question-and-answer interface that allows you to set up an entire Cisco Unified CallManager Express system at one time, Use the telephony-service setup command to start the Cisco Unified CallManager Express setup utility. If you do not use the setup keyword, you can set up phones one at a time using router CLI. The setup keyword is not stored in the router NVRAM. Note Ifyou attempt to use the automated setup option for a system whose telephony-service configuration is not empty, an error message advises you to remove the existing configuration first by using the no telephony-service command Prior to running the automated setup utility, configure the Cisco Unified CallManager Express router with Network Time Protocol (NTP) and load the appropriate firmware files into flash memory on the Cisco Unified CallManager Express router. The actual configuration is created only when the entire question-and-answer dialog has been completed. You can interrupt the process by pressing Ctrl-C at any point prior to the final question without having any configuration occur. The first question asked by the automated setup utility deals with DHCP and whether the Cisco Unified CallManager Express router will be providing this service. If you enter “y,” you must enter the parameters of the DHCP scope when the setup utility prompts you to do so. Entering “n" will skip the configuration of DHCP. The name of the scope that is automatically created if “y" is answered is “ITS.” Second, the automated setup utility configures the telephony service. The setup utility asks if the telephony service should be started. If you enter “y,” when prompted to do so, the IP address and port that Cisco Unified CallManager Express runs on will need to be entered. The IP address that you enter should be the address on the LAN that is local to the IP phones. The phones register with this address. In most cases, you should leave the port set to the default port of 2000. Entering “n” will stop the configuration of Cisco Unified CallManager Express. 182 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Third, you must choose the number of phones to configure. Specify no more than the licensed amount. If you choose fewer than the licensed amount, you can manually add more ephones later. Fourth, you are asked if dual lines are desired. If you enter “y,” the phones are configured like PBX phones; if you enter “n,” the phones are configured similar to keyswitch phones, The fifth question deals with the language of the phones and configures the locale that the IP phone will display. This step includes configuration of the SCCP-dictionary.xml and phonemodel-dictionary.xml files. Running the Automated Setup Utility (Cont.) When the configuration [ch Cal Proares is committed, the {eee settings selected 2 Germany appear in the running 3 Russia configuration ‘Sp 6 Netherlands 7 Norway 8 Portugal 9uK 10 Denmark ‘Whats the fst extension number you want to configure: [0]: 9000 Do you have Directinward-Dial service forall your hones? [yesinol:y Enter te full E164 number forthe fist phone 2095559000 Do you want to forward calls to a voice message sevice? (yesinol:y Entr extension or plot numberof the voice message service:0909, Call forward No Answer Timeout (1:10 ‘you wish to change any ofthe above information? [yesino|: ® The next part of the automated setup configures the call progress tones on the IP phones. The call progress tones are the sounds a caller hears. These include the dial tone, busy signal, ringback, and reorder signal. These call progress tones vary from country to country and should be set according to the tones that users are accustomed to hearing. To continue the automated setup, enter the first of the directory numbers that will be assigned. The system will assign the directory numbers in sequential order. If direct inward dialing (DID) needs to be set up, enter “yes” when prompted. DID numbers are used when the connection to the public switched telephone network (PSTN) is able to pass the dialed number. In order for this to happen, the connection should be ISDN. If the connections are Foreign Exchange Office (FXO), then a private line, automatic ringdown (PLAR) on the analog trunk must be set up instead. This configuration must be done manually—it is not included in the automated setup. Setting up DID can be very simple, especially if there is a relationship between the PSTN number and the internal directory number (for example, if 209 555-9009 maps to 1009). If there is no common relationship between the PSTN number and the internal directory number, then manual setup is required (for example, if 209 555-9009 maps to 7691). © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-183,The next question asks whether calls should be forwarded to a voice message service. Assuming that there is a voice mail system, you must enter the pilot point number. This sets “forward no answer” and “forward busy” to the pilot point number for all phones created. The timeout value for “forward no answer” also needs to be set; 18 seconds is the default. This value is in seconds rather than number of rings because ring lengths can vary by as much as 2 seconds. The final question in the setup utility asks if any of the information that was entered needs to be changed. If you enter “y,” the setup starts over. If you enter “n,” the changes are committed to the running configuration. One more step is required because the automated setup does not save the configuration at the end. Use the copy running-config startup-config command to save your setup configuration. Example: Complete Automated Setup ‘DHGP pool erated ‘Betwork 10,90.0.0 255.255.255.0 ieee s1aans790403020214.085 for frmare ever foung, it wilbe telephony-aervice leaded > 100d 7910 Po0403020216 AIL Fes creat ond 7960-7540 790309020214 tbootup sna’ > create ont-tilee Toaded to RAM. Firmware avaiable foTPtP server ‘ound, this entry Ismade igure shows the results of an automated setup, Note that the automated setup assumes that there is only one ephone-dn per ephone. Note “ITS* was the original name of Cisco Unified CallManager Express and still appears in some configurations. “184 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc.Quick Configuration Tool This topic describes the Cisco IP Communications Express Quick Configuration Tool (QCT). Overview of Quick Configuration Tool Web-based question and answer Designed for quick deployment Creates Cisco 10S commands in the background Assumes router is in factory default configuration ~ Can reset router to factory default if needed + Generates a configuration file + Uploads the configuration to the router Using the QCT application, a Cisco reseller partner or system administrator can dramatically reduce the time required to initially configure an IP telephony solution based on Cisco Unified CallManager Express and Cisco Unity Express, as compared to manually performing the configuration of Cisco Unified CallManager Express using the CLI of the Cisco IOS software. Afier downloading the QCT application, a system administrator can easily complete an IP telephony configuration based on Cisco Unified CallManager Express for a small or medium- sized business (SMB) with 50 phones or fewer. ‘The QCT application, through its user-friendly web-based GUI, prompts the administrator for the common parameters needed to configure the Cisco integrated services router to support a complete IP telephony solution. It gives the user the choice of configuring Cisco Unified CallManager Express in either a PBX mode with direct inward dialing (DID) extensions or a key system square mode. QCT supports the following telephony configuration parameters: = Basic network configuration: Network Time Protocol (NTP) configuration, TFTP, and DHCP options = Basic telephony features: Usernames, extensions, IP phone types and MAC addresses, public switched telephony network (PSTN) parameters, basic voice mail, and dial peers for 911 and 9+ calls = Advanced telephony features: Paging, intercom, Call Park, hunt group, and caller ID blocking QCT also supports automatic discovery of the hardware setup of the Cisco router, so the user can deploy an IP telephony configuration to a Cisco router without having detailed knowledge of the modules installed in the router. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-11Quick Configuration Tool ur to ao!9 dtu com Cofiguied Huet Name CMERouter Parcvord [eisco ompany Name: Cisco How Many P Phoree gong tobe D Aainoisiater set PDX Tene Zone GMT-8 (Pace) ¥ Sine Geneatad Configuston othe Start-Up cong on This figure shows use of QCT to configure host name, administrative username, password, tim zone, and number of IP phones. home on Cisco 38 pansion sit (AlN ‘appropriate interac dul that are inetalied, Slot n fect Harehave Cariguration Cisco 3828 * Ts Platform Supports us 19168 IF Phones a [eMery EMPTY - 2 THEI ~ 3 EMPTY vic 0 EMETY vac o: EMPTY This figure shows the hardware detection that allows users to configure an IP telephony system without having detailed knowledge of the Cisco router hardware architecture, expansion modules, of slot configurations. You can also set the configuration manually.Quick Configuration Tool (Cont.) ‘This figure shows configuration of Cisco Unified CallManager [5 like a PBX or keyswitch, specifying the first extension number, DHCP settings, NTP server, and VLAN configuration. ick Configuration Tool (Cont.) Chaco Unty Exess © This figure shows the PSTN configuration and voice mail integration with Cisco Unity Express. {© 2006 Cisco Systems, ne Cisco Unified CalNtanager Express Configuraton 2187Quick Configuration Tool (Cont) This figure shows the configuration of the IP phones. 2188 IP Telephony Express (IPT) wt.0 (© 2006 ciseo Systems, ncQuick Configuration Tool (Cont.) ‘lek Bowe to ost he consgurton fon yous compute Chek Upload 1 apace he contgratic ef the axe Longuatin File When the configuration tasks have been completed, QCT can connect to Cisco Unified CallManager Express to configure the route. {© 2006 Cisco Systems, ine. Cisca Unified Callvanager Express Gonfiguraton 2-1882190 Optional Parameters This topic identifies optional IP phone parameters. eV ar tetas + Language of phone display + Locale for call progress tones and cadences Ue You can customize the Cisco Unified CallManager Express system to some degree with the local language on the IP phone, call progress indicators, and cadence. This customization allows users to hear and interact with the system using the language and audible cues that are familiar to them, You can modity the Format in which the phone displays the date and time to the format that is typical forthe location of the installation IP Telephony Express (PTX) v4.0 (© 2006 Cisco Systems, IneRouter Configuration for Locale Parameters cMBRouter (conf ig-telephony-service) # user-locale language-code + Specifies the language to be displayed on an IP phone cusRouter (config-telephony-service) # network-locale language-code + Specifies the set of call progress tones and cadences on the IP phone On the Cisco Unified IP Phone 7940G and the Cisco Unified IP Phone 7960G, the language that is displayed and the call progress tones and cadences can be set to one of several ISO 3166 codes that indicate specific languages and geographic regions. Note The Cisco Unified Wireless IP Phone 7920 supports English, French, German, and Spanish, and you configure this setting on the handset. The user-locale and network-locale commands have no effect on the Cisco Unified Wireless IP Phone 7920. To see which language codes the user-locale command supports on your device, enter the following command CMERouter (config-telephony) #user-locale ? The following is a list of typical language codes supported: = DE: Germany DK: Denmark ES: Spain FR: France IT: Italy NL: Netherlands NO: Norway PT: Portugal RU: Russian Federation = SE: Sweden @ US: United States ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-19m= JA: Japan To see which language codes the network-locale command supports on your device, enter the following command: (CMBRouter (config-telephony) #network-locale ? The following is a list of typical language codes supported: = AT: Austria m CA: Canada = CH: Switzerland = DE: Germany = DK: Denmark = ES: Spain m= FR: France = GB: United Kingdom = IT: Italy = JA: Japan = NL: Netherlands = NO: Norway = PT: Portugal = RU: Russian Federation = SE: Sweden = US: United States Note ‘Changes to the language or call progress tones require that you reset the Cisco IP phone. 2-192 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IncDate and Time Parameters cMsRouter (config-telephony-service) # date-format (mm-dd-yy | ad-mm-yy | yy-dd-mm | yy-mm-dd) + Sets the date format for IP phone displays cusrouter (config-telephony- service) # time-format (12 | 24} + Selects a 12-hour or 24-hour clock for IP phone displays On the Cisco Unified IP Phone 7940G and the Cisco Unified IP Phone 7960G, the date and time format can be set on a systemwide basis for all IP phones. To see which date formats your device supports, enter the following command: CMERouter (config-telephony) #date-format ? The following is a list of typical date formats supported: = dd-mm-yy: Sets date to dd-mm-yy format = mm-dd-yy: Sets date to mm-dd-yy format = yy-dd-mm: Sets date to yy-dd-mm format = yy-mm-dd: Sets date to yy-mm-dd format To see which time formats your device supports, enter the following command: CMERouter (config-telephony) #time-format ? The following is a list of typical time formats supported: 12: Sets time to 12-hour (a.m. and p.m.) format = 24: Sets time to 24-hour format © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-193Rebooting Cisco Unified CallManager Express Phones This topic discusses rebooting IP phones. Rebooting with the reset and restart Commands reset Command restart Command Hard reboot + Soft reboot Phone firmware changes + Phone button changes User locale changes + Phone line changes Network locale changes + Speed dial number changes URL parameter changes + DHCP and TFTP not invoked DHCP and TFTP invoked More time-consuming than restart After you update information for one or more phones associated with a Cisco Unified CallManager Express router, you must reboot the phone or phones. There are two commands for rebooting: reset and restart. The reset command performs a hard reboot that is similar to a power-off, power-on sequence. It reboots the phone and contacts the DHCP server and TFTP server to update from their information as well. The restart command performs a soft reboot by simply rebooting the phone without contacting the DHCP and TFTP servers. The reset command takes significantly longer to process than the restart command when you are updating multiple phones, but you must use it after updating firmware, user locale, network locale, or URL parameters. For simple button, line, or speed dial changes, you can use the restart command. Use the reset command in ephone configuration mode to perform a complete reboot of a single IP phone. This command has the same effect as a reset command in telephony-service mode that you use to reset one phone or all phones. 2-194 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc.reset Command Configuration cuBRouter (config-telephony-service) # reset (all (time-interval] | cancel | mac-address sequence-all} + Resets one or all phones cupRouter (conf ig-ephone) # reset + Resets a specific ephone To perform a complete reboot of one or all phones associated with a Cisco Unified CallManager Express router, use the reset command in telephony-service configuration mode. When you are using the reset command from telephony-service mode, the default time interval of 15 seconds is recommended for an 8- to 10-phone office so that the phones do not attempt to access TFTP server resources simultaneously. You should increase this value for larger networks. When you use the reset sequence-all command, the router waits for one phone to complete its reset and reregister before starting to reset the next phone. The delay provided by this command prevents multiple phones from attempting to access the TFTP server simultaneously and therefore failing to reset properly, Each reset operation can take several minutes when you use this command. There is a reset timeout of 4 minutes, after which the router stops waiting for the currently registering phone to complete registration and starts to reset the next phone. If the router configuration is changed so that the XML configuration files for the phones are modified (changes are made to user locale, network locale, or phone firmware), then ‘ whenever the reset all or restart all command is used, the router automatically executes the reset sequence-all command instead. The reset sequence-all command resets the phones one at a time to prevent multiple phones from trying to contact the TFTP server simultaneously. This one-at-a-time sequencing can take a long time if there are many phones. To avoid this automatic behavior, use the reset all time-interval command or the restart all sime-interval command and set a time interval that is not equal to the 15-second default time interval (for example, set a time interval of 14 seconds). If you have started a reset sequence-all command in error, use the reset cancel command to interrupt and cancel the sequence of resets To perform a complete reboot of a single phone associated with a Cisco Unified CallManager Express router, use the reset command in ephone configuration mode. ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-195restart Command Configuration cusrouter (config-telephony-service) # restart {all (time-interval] | mac-address} + Restarts one or all phones cMsRouter (config-ephone) # restart + Restarts the ephone The restart command causes the system to quickly perform a phone reset in which only the button template, lines, and speed dial numbers are updated. This command is much faster than the reset command because the phone does not access the DHCP or TFTP server. For updates related to phone firmware, user locale, network locale, or URL parameters, use the reset command, To restart a single phone, use the restart command with the mac-address argument or use it in ephone configuration mode. -196 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc.Setup Troubleshooting Tips This topic identifies IP phone setup troubleshooting tips. Setup Troubleshooting Overview Verify that the correct IP address and scope options are received on the IP phone. Verify that the correct files are in flash memory. Debug the TFTP server. Verify the installation of the phone firmware. Verify that the locale is correct. Verify phone setup. + Review the configuration. With automated setup, there are many places to check if you encounter problems. Some of the more usefull places to check and tools to use include the following: Verify the IP addressing: Use the Settings button to check the configuration on the IP phone. Verify the files in flash memory: Check and verify that the correct firmware files are present in flash memory Debug the TFTP server: Make sure that the firmware and XML files are being served correctly. Verify the firmware installation of the phones: Use the debug ephone register command to verify which firmware is being installed. Verify that the locale is correct: Use the telephony-service tftp-bindings command to view the files being served up by the TFTP server. Verify the phone setup: Use the show ephone command to view the status of the ephones and whether they are registered correctly Review the configuration: Use the show running-config command to verify the ephone- dn configuration, ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-197Verifying IP Addressing + Use the Settings button and choose Network Configuration. + Verify that the IP address and subnet mask are correct. + Verify that the TFTP server is the Cisco Unified CallManager Express router. + Verify that the default gateway is correct. To verify that the DHCP server is providing the correct information to the IP phones, press the Settings button, then choose Network Configuration, Scroll through the settings and verify the IP address, subnet mask, default gateway, and location of the defined TFTP server. The TFTP server must be the Cisco Unified CallManager Express router. Verifying Correct Firmware Files in Flash show flash Command length —-~--date/time-----~ path ‘The show flash command displays the contents of flash memory. The flash memory must contain the firmware files that are necessary for the IP phone models that are deployed. Many other files may be here as well, depending on other configurations. 2-198 IP Telephony Express (IPTX) v4.0 ‘© 2006 Cisco Systems, Inc.lebug tftp events Command + Opened system: /ste/RWDetauie7960-cnf xml, £4 0, size 784 for ‘Opened system: /ste/2WLDetasit7960-cnf sal, £4 0, size 784 for Pinsaned ayatam:/tt8/KnDeeaule7960.enf mi, tine 00:00:00 for Pinsahed system:/tte/mpetaule7960.0n€ sa, time 0070000 for + Can verify that the SEP file for the phone is found + Can verify that the correct firmware has been downloaded The debug tftp events command enables the administrator to view output regarding files that are served up by the TFTP server. The administrator can view files, including firmware, that are specil ¢ to Cisco Unified CallManager Express to see whether out-of-date or unsupported files are being used. The administrator can also view the XML files for configured IP phones, the XML files for new IP phones, and locale files. If the firmware ends with the .bin extension, then the file is unsigned. If the firmware ends with the .sbin extension, then the file is signed. If the .sbin extension is used, the IP phone permanently requires signed firmware loads and cannot use unsigned firmware loads © 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-199Verifying Phone Firmware Installation Mer 2 15:16157.582! Mew skinny socket accepted (2) (2 active) mar 2 18:16:57,765 jormacional pisi06e (0x30) p2-184S8i946 (OxB000A08), war 2 isdess7.766 (1) StationRagiaterstensage (1/2/2) from 30-50-0.21 Mar 2 isst6ss7.766: ophone-(1} {1} dtationddentifier Inevance 1 deviceType 7 Mar 2 28:26:57,766: ephone-L{-1l retationZpAddr 10.90.0012 war 2 15426;57.768+ ephone-1(2] -phone SRFOOOFZA7070F@ ve-azoctate OX on aocket (2) ar 2 15426457.768: ephone-203) wore Yar 2 15;18557.766: sphone-A(A](SRFOOOFSA7OPOTS] Reghaterhck sont to ephone 1: bewpalive Verify the correct phone firmware installation by setting registration debugging with the debug ephone register command, Then reset the phones and look at the Skinny Station larmMessage displayed during phone reregistration. The Load= parameter should appear in the display, followed by an abbreviated version name that corresponds to the correct firmware filename. Verifying Locale-Specific Files rover syatem:/ita/SRPDRFAULT cnt alias SEPDetault cnt evar yatem:/ita/1LDefault cnt enl alias 2¥Default.onf aml eft. in: /ite/united statee/1960-tones.aml alias United staten/7960-tonea xml Use the show telephony-service tftp-bindings command to ensure that the locale-sp files are correct. 2-200 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Verifying Cisco IP Phone Setup Enter the show ephone command to verify the Cisco IP phone setup after the phones have registered with the Cisco Unified CallManager Express router. '© 2006 seo Systems, no. Cisco Unified CallManager Express Configuration 2201Verifying Cisco Unified CallManager Express Phone Configuration This topic describes how to verify the Cisco Unified CallManager Express configuration, ACV Reem emer Lele (id Express Phone Configuration A PR Use the show running-config command to verify the configuration. The primary area of the TFTP configuration, the ephones, and the ephone-dns. merest for Cisco Unified CallManager Express functionality is the telephony-service section, 2.202 IP Telephony Express (PTX) v4.0 18 2006 Cisco Systems, ne.Summary This topic summarizes the key points that were discussed in this lesson. 1d CallManager Express requires firmware files to bbe copied to the flash memory on the router and shared using TFTP, + There are four ways to create a phone setup in Cisco Unified CallManager Express: manually, partially automated, automated, and using QCT. + Manual setup can save time and effort when used for multiple-site deployments. + Partially automated setup allows for the deployment of many phones without the work of configuring every phone ‘manually. + QCT can dramatically reduce the time it takes to configure an IP telephony solution based on Cisco Unified CallManager Express and Cisco Unity Express. a gENRNRE aT RE es fa Ea Summary (C + The Cisco Unified CallManager Express setup utility provides ‘a question-and-answer interface. + IP phones can be rebooted using the reset and restart command, + When troubleshooting, there are many show and debug commands available. + You can use the show running-config command to verify the configuration. 185 - iE le {© 2008 Cisco Systems. Inc. Cisco Unified Galanager Express Configuration 2-208,Module Summary This topic summa izes the key points that were discussed in this module. Module Summary + Cisco Unified CallManager Express solutions are based on Cisco 10S software and can be managed either through the CLI or a GUL. Cisco Unified CallManager Express Release 4.0 requires a minimum Cisco 10S Software Release 12.4 (9). The network configuration and services that are required by Cisco Unified CallManager Express include proper switch configuration, DHCP, and NTP. + The IP phone or analog device needs the following files to successfully register with the Cisco Unified CallManager Express router: firmware, XMLDefault.cnf.xml, and SEPAAAABBBBCCCC.cnt xml. + Transcoding resources need to be configured when a Module Summary (Cont.) + The Cisco Unified CallManager Express system can be configured in various ways by using ephones and ephone- dns in different ways. The ephone represents the configuration and settings of the physical phone, and the ephone-dn represents a line or channel for voice to connect to the ephone. + Cisco Unified CallManager Express can be configured to support SIP endpoints. + The Cisco Unified CallManager Express system can be deployed using the following methods: automated, partially automated, using QCT, and manually. '© 2006 Cisco Systems, ne Cisco Unified CallManager Express Contiguraton 2-205References For additional information, refer to the following resources: Cisco Systems, Inc. Cisco Unified CallManager Express 4.0(0) Supported Firmwar Platforms, Memory, and Voice Products. htip://www. cisco. com/en/US/customerproducts/sw/voicesw/ps4625/prod, £091 860080515908. html Cisco Systems, Ine, Configuring DHCP. htapzi/www.cisco.com/univered’ce'tddoe!productsoftwarelios122/122cger/fipr_efipeprt 1 cefudhep.htmxtocidd. Cisco Systems, Inc, Cisco Unified CallManager Express System Administrator Guide. hutp://www.cisco.com/en/US/products’sw/voicesw/ps4625/products_configuration_guide_b 00091 86a00806a80dd, htm] Cisco Systems, Inc. Cisco CallManager Express 3.2: Setting Up a Cisco CallManager Express System tip: /cisco.comven/USpartner/products/sw iosswrel/ps5207/product er 9186200802d253¢ htm. NTP Project. NIP: The Network Time Protocol. http:/wwww.ntp.org. Cisco Systems, Inc. Cisco Unified CallManager Express System Administrator Guide, “Transcoding Support.” hutp://www.cisco.com/en/US/products/sw/voicesw/ps4625/produets_configuration_guide_¢ hhapter091:86a00806a8015. htm! Cisco Systems, Inc. IP Phone Documentation for Cisco Unified CallManager Express 3.2 and Later. hutp:/‘www.cisco.com/univered eeltddociproductaccess/ip_ph/ip_ks/eme32/eme32uglinde xhim. Cisco Systems, Inc, Cisco Unified Cisco Unified CallManager Express Configuration Guide for SIP IP Phones. hutp://www.cisco.com/en/US/partner/products/sw/voiceswips4625/products_configuration_ ‘guide book09186a008052da6b htm Cisco Systems, Inc. Software Center (Downloads): Voice Software, Ittpy/www.cisco.com/kobayashi/sw-centerisw-voice.shtml (requires Cisco.com login), stallation_guid \ feature_guide_chapt 2206 IP Telepnony Express (PTX) v4.0 18 2006 Cisco Systems, ne.Module Self-Check Use the questions here to review what you leared in this module, The correct answers and solutions are found in the Module Self-Check Answer Key. Qn Q4) Qs) Which three are key features of Cisco Unified CallManager Express? (Choose three.) (Source: Understanding Cisco Unified CallManager Express Features and Functionality) A) built-in Auto Attendant with Cisco Unity Express B) interoperable with Cisco Unified CallManager Release 3.3 ©) supports HTML applications on IP phones, D) licensing upgradable to Cisco Unified SRST B) reduces TCO by converging voice, video, and data onto a common network F) — GUlor CLI administration CAC functionality is part of which Cisco Unified CallManager Express-supported protocol? (Source: Understanding Cisco Unified CallManager Express Features and Functionality) A) oRTP B) W323 ©) scce D) 4320 Unified IP phones are supported by Cisco Unified CallManager Express? (Choose three.) (Source: Understanding Cisco Unified CallManager Express Features and Functionality) A) ATA 188, B) 7920 c)7970G D) 79606 Which is one of the recommendations that Cisco makes for IP addressing deployment? (Source: Configuring Cisco Unified CallManager Express Network Parameters) A) Statically apply IP addresses to IP phones to ensure stability B) Apply public IP addresses to IP phones so that they can be reached from the PSTN. ©) Add IP phones with DHCP as the mechanism for obtaining addressees, D) Deploy IP phones on the same subnet as data devices. ‘What is the most efficient way to get multiple VLANs to the router? (Source: Configuring Cisco Unified CallManager Express Network Parameters) A) by using a high-speed Layer 2 switch B) by connecting a trunk directly between the IP phone and the router ©) by using the configuration known as “router on a stick” D) not possible with VLANs connected to IP phones (© 2006 Cisco Systems, Inc. (Cisco Unified CallManager Express Configuration 2.207Q6) Which is a reason to set up DHCP service for IP phones by defining a DHCP relay server? (Source: Configuring Cisco Unified CallManager Express Network Parameters) A) the Cisco Unified CallManager Express router is a DHCP server and you need different settings on non-IP phones on the same subnet B) the Cisco Unified CallManager Express router is a DHCP server with a single shared address pool for all your DHCP clients ©) the Cisco Unified CallManager Express router is not a DHCP server and you want to relay DHCP requests from IP phones to a DHCP server on a different subnet Q7) Which statement about the command router(dhep-config)# host jp-address subnet- ‘mask is correct? (Source: Configuring Cisco Unified CallManager Express Network Parameters) A) creates a scope of the entire subnet with the specified IP address in it B) __ is followed by assigning a host with a specific MAC address defined by the lient-identifier mac-address command ©) statically assigns an IP address to a host that would otherwise get it dynamically Q8) Why would a DHCP relay server need to be implemented? (Source: Cor Unified CallManager Express Network Parameters) ring Cisco A) __ ifthe DHCP server does not have a local interface on the network with the DHCP tie B) because the DHCP request and response process is not broadcast ©) to relay the proprietary DHCP request type of the IP phone to the standard DHCP request type understood by Cisco 10S software D) _ if an IP phone, a data device, and a DHCP server reside on the same subnet Q9) Which protocol and port does NTP use? (Source: Configuring Cisco Unified CallManager Express Network Parameters) A) TCP port 123 B) UDP port 123, ©) TCP port 213, D) UDP port 213, Q10) What do IP phones use during registration to download firmware files from the flash ‘memory of the router? (Source: Understanding the IP Phone Registration Process) A) HTTP B) DHCP c) FIP D) TFT 2208 IP Telephony Express (PTX) v4.0 (© 2006 Gis00 Systems, ne.Qi) Q12) Q13) Ql4) QIs) Q16) The use of the type command under the ephone phone type is required to register for (Source: Understanding the IP Phone Registration Process) A) __ the Cisco Unified IP Phone Expansion Module 7914 B) all valid IP phones other than the Cisco Unified IP Phone Expansion Module 7914 C) all ATA devices other than the Cisco Unified IP Phone Expansion Module 7914 D) _ no phones or devices, because the ephone can determine any of them automatically through the Cisco Unified CallManager Express system What is the first step in the process of an IP phone obtaining its XML configuration file and IP address? (Source: Understanding the IP Phone Registration Process) A) The switch applies power to the line, B) The powered device has a physical link when there is no power between the pin that the FLP arrives on and a pin that goes back to the switch. This creates acireuit, and the end result is that the FLP arrives back at the switch. This never happens when the device attached is not a powered device, like a PC. As a result, if the FLP does not return to the switch, no power is applied C) The switch sends a special tone called an FLP out the interface, and this FLP goes to the powered device, which in this case is an IP phone. D) The switch applies power to the line. Which command creates an ephone-dn that builds one virtual voice port? (Source: Defining Ephone-dn and Ephone) A) router(config-ephone-dn)# ephone-dn dh-tag B) _router(config-ephone-dn)# number dn-number C) _router(config}# ephone-dn dn-tag D) _router(config)# ephone-dn da-number Which is the first command to create or modify an ephone? (Source: Defining Ephone- dn and Ephone) A) ___router(config-ephone)# ephone phone-tag B) _ephone phone-tag from ephone subconfiguration mode ©) ephone phone-tag from global configuration mode Which two are types of ephone-dns that can be found in a Cisco Unified CallManager Express system? (Choose two.) (Source: Defining Ephone-dn and Ephone) A) single-line ephone-dn B) secondary and tertiary extension on one ephone-dn ©) shared ephone D) multiple ephones on one ephone-dn E) overlay ephone-dn Which command do you use to enable the registrar functions of Cisco Unified CallManager Express? (Source: Introducing SIP and Cisco Unified CallManager Express) A) CMERouter(config)#register sip to sip B) CMERouter(config-voi-sip)#registrar server C) _ CMERouter(config-voi-srv)#register server D) — CMERouter(voice-register-global)#registrar server sip to sip ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-209Q17) What does the command max-dn define? (Source: Introducing SIP and Cisco Unified CallManager Express) A) sets the maximum number of directory numbers on the Cisco Unified CallManager Express system B) sets the maximum number of directory numbers on the Cisco Unified CallManager Express system to 300 ©) sets the maximum number of directory numbers on an IP phone to the defined number D) sets the maximum number of directory numbers on an IP phone to the number of buttons on the IP phone QI8) Which three are true about using the voice register pool command? (Choose three.) (Source: Introducing SIP and Cisco Unified CallManager Express) A) The MAC address of the IP phone can be used to define the locally attached IP phone that is in the ARP cache of the Cisco Unified CallManager Express system, B) The MAC address of the IP phone can be used to define the remote IP phone that is defined in the Cisco Unified CallManager Express system. ©) Use the IP address of a locally attached IP phone with a static IP address. D) _Itis the subnet of an IP phone that is locally attached to the Cisco Unified CallManager Express system, E) Use the number of the first line on the IP phone that is cither locally or remotely attached, Q19) — What is the order of files that a Cisco IP phone checks when it boots? (Source: Introducing SIP and Cisco Unified CallManager Express) A) CTLSEP, SIP, SEP, SIPDefault, SEPDefault B) — CTLSEP, SIPDefault, SEPDefault, SIP, SEP c) SIP, SEP, SIPDefault, SEPDefault D) —_- CTLSEP, SEP, SIP, SEPDefault, SIPDefault E) —_ SIP, SEP, SIPDefault, SEPDefault, CTLSEP Q20) Which two does Cisco Unified CallManager Express use to share the firmware files that are copied to the flash memory on the router? (Choose two.) (Source: Describing Cisco Unified CallManager Express Files) A) HTTP B) TCP Cc) FTP D) —‘TFTP E) cpp Q21) Which file bundle contains all the files that are needed to run the GUI web interface for Cisco Unified CallManager Express and Cisco Unity Express? (Source: Describing Cisco Unified CallManager Express Files) A) CiscolOSTSP.zip B) —_eme-b-acd-2.0.0.0.tar ©) eme-gui-123-11XL.tar D) ——_xmltemplate 0 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.Q22) Q23) Q24) Q25) Q26) Q27) Q28) Which file bundle contains all the files that are needed to allow third-party software to interact with the Cisco Unified CallManager Express system through TAPI Lite? (Source: Describing Cisco Unified CallManager Express Files) A) CiscolOSTSP.zip B) —_cme-b-aed-2.0.0.0.tar C) —— eme-gui-123-11XL.tar D) —_xml.template Which file is a sample file for creating a customer administrator with a limited subset of administrative privileges? (Source: Describing Cisco Unified CallManager Express Files) A) music-on-hold.au B) — cme-gui-123-11XL.tar Co xml.template Which is the maximum number of ephone-dns and ephones supported by the service before configuring the telephony service? (Source: Understanding Initial Phone Setup) A) 0 B) 100 C) 288 D) unlimited Which command is used to perform an automated phone setup in a Cisco Unified CallManager Express system? (Source: Understanding Initial Phone Setup) A) router(config)# telephony-service setup B) _router(config-telephony-service)# telephony-service setup C) __router(config)# auto assign start-dn to stop-dn D) __router(config-telephony-service)# auto assign start-dn to stop-dn Which ephone-dns can automatically assigned ephone-dn tags belong to? (Source: Understanding Initial Phone Setup) A) paging ephone-dns B) _ intercom ephone-dns C) MOH ephone-dns D) — MWlephone-dns E) normal ephone-dns On which phone is the language setting made on the handset rather than by using the user-locale and network-locale Cisco IOS commands? (Source: Understanding Initial Phone Setup) A) Cisco Unified Wireless IP Phone 7920 B) Cisco Unified IP Phone 7940G C) Cisco Unified IP Phone 7960G Which command performs a hard reboot, similar to a power-off, power-on sequence? (Source: Understanding Initial Phone Setup) A) restart B) reset Cc) either restart or reset ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 2-211Q29) Which procedure verifies that the DHCP server is providing the correct information to the IP phones?(Source: Understanding Initial Phone Setup) A) B) c°) D) issuing the show running-config command issuing the show flash command issuing the debug ephone register command pressing the Settings button, then, from the menu that appears, choosing the Network Configuration settings Q30) Which procedure verifies the Cisco Unified CallManager Express configuration? (Source: Understanding Initial Phone Setup) A) B) ° D) issuing the show running-config command issuing the show flash command issuing the debug ephone register command pressing the Settings button, then, from the menu that appears, choosing the Network Configuration settings 2-212 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, InModule Self-Check Answer Key Qn A B D TPOAOO>Tee>enNAAs >poRP>me>>aean B.D cD ‘© 2006 Cisco Systems, Inc. Cisco Unified CallManager Express Configuration 221Module 3 PSTN Interface and Voice Dial Peer Configuration Overview Cisco voice devices must support a wide variety of connection types. This module describes the function and basie configuration of various analog and digital voice connections. This module presents information on how to fine-tune voice ports with port-specific configurations and dial peers. This module also covers the use of digit manipulation and special-purpose connections, along with the Cisco implementation of supplementary telephony services. Module Objectives Upon completing this module, you will be able to configure analog voice interfaces, digital voice interfaces, and dial peers to set up VoIP communications. This ability includes being able to meet these objectives: m= Describe the various types of analog and di Cisco Unified CallManager Express al interfaces and signaling types supported by = Configure analog and digital voice interfaces and discuss voice port applications, FXS, FXO, E&M, BRI timers and timing, digital voice ports, CAS, and CCS and PRI = Describe dial peers and configuration tasks = Describe how call legs relate to inbound and outbound dial peers by defining all of the steps in the eall setup process and the proper use of digit manipulationLesson 1 Understanding Analog and Digital Voice Interfaces Overview Interfacing Cisco Unified CallManager Express with traditional analog telephony devi requires an understanding of the various interfaces used in the industry. When you require additional port density and features, you can use a digital connection. This lesson describes the various analog and digital interfaces that you can use with Cisco Unified CallManager Express. ILalso explores analog and digital signaling between Cisco Unified CallManager Express and the central office (CO), as well asthe various forms of connection. The choice of digital connection can vary based on cartier, and not all services may be available in all are es Upon completing this lesson, you will be able to identify and describe the various analog and digital interfaces and signaling types supported by Cisco Unified CallManager Express. This ability includes being able to meet these objectives: = Identify the components of local-loop connections = Describe FXS, FXO, and E&M interfaces = State the uses and types of CAS systems that are used for T1 State the uses and types of CAS systems that are used for E1 = State the uses and types of CCS systems = Describe what PRI and BRI are and how they can be usedLocal-Loop Connections This topic describes the parts of a traditional telephony local-loop connection between a telephone subscriber and the telephone company Local-Loop Connections Telephone Switch A loop is the pair of physical wires from the subscriber to the telephone company switch. A subscriber home telephone connects to the telephone company CO via an electrical communication path called a “local loop.” as depicted in the figure, The loop consists of a pair of twisted wires—one is the tip wire, and the other is the ring wire. ‘In most arrangements, the ring wire ties to the negative side of a power source, the battery, and. the tip wire connects to the ground. This pair of wires, which represents the local loop, along with all of the other pairs in your neighborhood, connects to the CO in a cable bundle that is either buried underground or strung on poles. When the analog phone or fax goes into the off- hook state, an electrical circuit is completed and current flows through the loop. This current signals the switch that the analog phone or fax is off-hook. The switch then uses a dial tone ‘generator fo send a signal—the dial tone—that the switch is ready to receive digits 34 IP Telephony Express (IPT) 0 (© 2006 Cisco Systems, IneAnalog Voice Interfaces This topic defines the three analog interfaces that you can install in a voice gateway: Foreign Exchange Station (FXS), Foreign Exchange Office (FXO), and ear and mouth (E&M). It also. discusses how each of these interfaces is used, Le Tar tot &y FXS, FXS “y FXS + Connects directly to analog phones or faxes + Used to provision local service + Provides power, call progress tones, and dial tone a ‘When you use analog phones or fax machines in an IP-based environment, they must have a connection into this IP network. This connection takes the form of an FXS interface. The FXS. interface provides a direct connection to an analog telephone, a fax machine, or a similar device. From the perspective of the analog device, the FXS interface functions like a switch Therefore, it must supply line power, ring voltage, and dial tone. “The FXS interface contains the coder-decoder (codec), which converts the spoken analog voice ‘wave into a digital format for processing by the voice-enabled deviee. Note ‘You cannot forward analog phones that you plug into an FXS port on the Cisco Unified CallManager Express router to Cisco Unity Express voice mail If you need voice mail on the ‘analog phones, use the Cisco ATA 186 or the Cisco ATA 188 Analog Telephone Adaptor to Connect the analog phone to the network {© 2008 Gisco Systems, in PSTN inerface and Voice Dial Peer Configuration 3Sbp CO} aii (er Connects directly to office equipment + Used to make and receive calls from the PSTN + Can be used to connect through the PSTN to another site Answers inbound calls In order for standard analog connections from the CO to enter the IP network, they must terminate on an interface on a voice gateway. You can use an FXO interface for this purpose. When a call arrives, the FXO interface answers the call and either presents a second dial tone or is configured with private line, automatic ringdown (PLAR). For outbound calls, the FXO interface provides either pulse digits or dual tone multifrequency (DTMF) digits for outbound dialing 16 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, Inc.E&M Interface Connects two sites with a leased connection Allows for the use of non-PSTN numbers Used to create tie-lines Commonly used to connect to external music on hold sources You can lease special analog connections called “tie-lines” from the carrier. You typically use these to tie together two or mores sites that have analog connections. This tie-line terminates in an analog interface on the router so that the analog communication can enter the IP network. The E&M interface on the router is where you can terminate these tie-lines. E&M signaling is also referred to as “recEive and transMit’; it comes from the term “earth and magneto.” “Earth’ represents the electrical ground, and “magneto” represents the electromagnet used to generate tones. E&M signaling defines a trunk-circuit side and a signaling-unit side for each connection, similar to the DCE and DTE reference types. The router is usually the trunk-circuit side, and the telephone company (telco), a CO, a channel bank, or a Cisco voice-enabled platform is the signaling-unit side Note Many music on hold (MOH) services provide an analog E&M interface that you can use to connect to the Cisco Unified CallManager Express router. ‘© 2006 Cisco Systems, Inc. PSTN Interface and Voice Dial Peer Configuration = 3Channel Associated Signaling Systems: T1 This topic describes channel associated signaling (CAS) and its uses with TI transmission. Channel Associated Signaling Systems Extonded Super Frame Audio Address Signaling (OTM) LTT I HT Because the signaling occurs within each digital service level 0 (DSO), it is referred to as in- band. And because the use of these bits is reserved exclusively for signaling each respective voice channel, it is referred to as “channel associated signaling,” or CAS. Super Frame (SF) has a 12-frame structure and provides A&B bit signaling, Extended Super Frame (ESF) has a 24-frame structure and provides ABCD signaling. ‘The audio path can carry tones, such as DTMF addressing or call progress tones. However, other CAS signals must be carried via the robbed bits. These robbed bits are the least significant bits in the audio channel. 38 IP Telephony Express (IPTX) v4.0 © 2006 Cisco Systems, IncCharacteristics of a CAS T1 + Up to 24 channels for voice + Each channel is a DSO + 8000 samples per second + 1 byte per sample + Partial T1 may be available + Signaling travels in-band You can connect Cisco Unified CallManager Express to the public switched telephone network (PSTN) through a CAS TI connection. This configuration provi Each channel is a 64-kbps DSO. es up to 24 channels for voice. ‘© 2006 Cisco Systems, Inc. PSTN Interface and Voice Dial Peer Configuration 34Channel Associated Signaling Systems: E1 This topic describes the uses of CAS with El transmission, Channel Associated Signaling Systems: E1 Eee} | EO Nos Contains Only ‘Signaling Information (Time Siot No.1) (Time Stot No.17) In El framing and signaling, 30 of the 32 available channels, or time slots, are used for voice or data, Framing information uses time slot | (channel 0), an time slot 17 (channel 16) is used for signaling by all the other time slots. This signaling format is also known as CAS because each bearer channel has specific bits in the [7th time slot that are assigned for signaling. However, Cisco considers this implementation of CAS as out-of-band because the voice channel does not carry the signaling bits within the voice channel, as is the case with TI Note E1 circuits do not use robbed-bit signaling 3:40 IP Telephony Express (PTX) v4.0 (© 2006 Cisco Systems, IncCharacteristics of a CAS E1 CAS E1 Up to 30 channels for voice Each channel is a DSO + 8000 samples per second 1 byte per sample Partial E1 may be available + Signaling is carried out-of-band You can connect Cisco Unified CallManager Express to the PSTN and it can provide up to 30 channels for voice. 1© 2006 Cisco Systems, ne PSTN interlace and Voice Dial Peer Configuration 3-11Common-Channel Signaling Systems This topic describes common-channel signaling (CCS) systems. Common-Channel Signaling Et Te Siot 17 “ine Slot 17 | | I | ZN © Ele rn Supervision Adaress secress GnoR Hook Sigating Signing (ois Puicey ‘or pac eS SN While CAS uses bit time slots assigned to each specific channel, CCS uses a common channel and protocol to set up calls forall the bearer (B) channels, For example, for ISDN over El, the signaling protocol Q.931 uses time slot 17 to exchange call setup messages for any of the 30 B channels. Examples of CCS are as follows = Proprietary implements Some PBX vendors choose to implement a proprictary CCS protocol between their PBXs for TI and El. In this implementation, you configure Cisco devices for Transparent Common Channel Signaling (T-CCS) because they do not understand proprietary signaling information and must simply transport the signaling, ‘without modification or interpretation, = ISDN: Uses the Q.931 signaling protocol in a common channel to signal all other channels, ‘© Digital Private Network Signaling System (DPNSS): This open standard was developed by British Telecom for implementation by any vendor that chooses to use it. DPNSS also uses a common channel to signal all other channels © Q Signaling (QSIG): Like ISDN, QSIG uses a common channel to signal all other channels 342 IP Telephony Express (IPT) vs 0 '© 2006 Cisco Systems, nc.PRI and BRI This topic describes PRI and BRI and how you can use them to support voice. LSP] E agi.) PRI23B+D €3 = BRI2B+D eB + Allows for multiple services through one connection + Well adapted for voice ~ 64-Kkbps channels Q.931 on the D channel + Supports standards-based functions + Supports proprietary implementations, + International utiiz Eee ISDN is one form of CCS. PRI and BRI are the two ways of implementing ISDN. Note Because ISON is a digital service, the time required to set up a callis significantly less than the time required to set up an analog call PRI supports 23 (for T1) or 30 (for El) B channels, while BRI features two B channels. Each implementation also supports a single data (D) channel that is used to carry signaling information. The following are characteristics of ISDN PRI and BRI = ISDN channels can carry data, voiee, or video. = Each B channel is 64 kbps, and G.711 pulse code modulation (PCM) requires 64 kbps, so this is a perfect match for voice applications. m= The D channel in BRI is 16 kbps and in PRI is 64 kbps. = ISDN has a built-in call control protocol known as ITU-T Q.931 that runs on the D channel = ISDN can support standards-based voice features, such as call forwarding, and standards- based enhanced dialup capabilities, such as Group TV fax and audio channels. = ISDN can carry vendor-specific PBX features, ISDN BRI voice is commonly used in Europe; ISDN PRI voice is used worldwide, (© 2006 Cisco Systems, Inc. PSTN Interface and Voice Dial Peer Configuration 3.13Summary This topic summarizes the key points that were discussed in this lesson. RTT ag + The local loop consists of a pair of twisted the tip wire, and the other is the ring wire. + Analog interfaces, such as FXS, FXO, and E&M, can be used to connect analog devices and to connect to the PSTN. ied CallManager Express can use T1 circuits to + Cisco Unified CallManager Express can use E1 circuits to convey voice. + Examples of CCS are proprietary implementations, ISDN, DPNSS, and QsiG. + ISDN can be implemented in two ways: BRI and PRI. 3:14 IP Tolophony Express (IPTX) v4.0, {© 2006 Cisco Systems, IneLesson 2 Configuring Analog and Digital Voice Interfaces Overview The connections to analog devices—the public switched telephone network (PSTN) and WAN links between sites—may take either an analog or a digital form. The analog interfaces that are ‘commonly found include Foreign Exchange Station (FXS), Foreign Exchange Office (FXO). and ear and mouth (E&M) interfaces. You use an FXS interface to connect analog devices, such as phones and fax machines. You typically use FXO interfaces for traditional analog ‘connections to the PSTN. You typically use E&M analog connections for connections to the PSTN, and you can use them for analog tie-line connections to another site or to connect a music on hold (MOH) system. ‘The digital connections include both channel associated signaling (CAS) and common-channel signaling (CCS) digital connections. The CAS connection has signaling in-band. This means that the voice and the signaling travel together on the same circuit, CCS links use out-of-band signaling. The most common form of CCS is the ISDN service. There are two main offerings in ISDN: BRI and PRL To connect to an ISDN network, you must use the correct router interface, BRI requires specific commands to enable ISDN. You typically use ISDN BRI for remote access at small branch sites with lower bandwidth requirements. Larger central sites with higher bandwidth requirements typically use PRI to aggregate multiple BRIS. Internet service providers (ISPs) also use ISDN PRI to support large numbers of plain old telephone service (POTS) (analog modem) and ISDN BRI calls.Objectives Upon completing this lesson, you will be able to configure analog and digital voice interfaces. This ability includes being abie to meet these objectives: = Set the configuration parameters for FXS voice ports = Set the configuration parameters for FXO voice ports = Set the configuration parameters for E&M voice ports = Set timers and timing requirements on ports to adjust the time allowed for specific functions = Set the configuration parameters for digital voice ports = Set the configuration parameters for CAS voice ports = Set the configuration for BRI voice ports = Set the configuration parameters for PRI voice ports 3416 IP Telephony Express (IPT) vt 0 (© 2006 Cisco Systems, ne.Foreign Exchange Station Port Configuration EXS ports connect analog edge devices. This topic identifies the parameters that are configurable on the FXS port LPM LM oleate lle el led + signal + eptone + description + ring frequency + ring cadence + disconnect-ack + busyout + station id name + station id number aaa a In North America, the FXS port connection functions with default settings most of the time. ‘The same cannot be said for other countries and continents. Remember, FXS ports look like switches to the edge devices that connect to them. Therefore, the configuration of the FXS port should emulate the switch configuration of the local PSTN For example, consider the scenario of an international company with offices in the United States and England. The PSTN of each country provides signaling that is standard for that country. In the United States, the PSTN provides a dial tone that is different from the tone in England. And when the telephone rings to signal an incoming call, the ring in the United States is different from the ring in England, Another instance when you might change the default configuration is when the connection is a trunk to a PBX or key system, In that case, you must configure the FXS port to match the settings of that device. © 2006 Osco Systems, nc. PSTN Interface and Voice Dial Peer Contiguraion 3.17Configurat n Parameters FXS port configuration allows you to set parameters based on the requirements of the connection if you need to alter the default settings or you need to set the parameters for fine- tuning, You can set the following configuration parameters: ‘= signal: Sets the signaling type for the FXS port In most cases, the default signaling of loop start works well. Ifthe connected device is a PBX or a key system, the preferred signaling is ground start. Modem PBXs and key systems do not normally use FXS ports as connections to the network, but older systems may still have these interfaces. When connecting the FXS port to a PBX or key system, you must check the configuration of the voice system and set the FXS port to match the system setting, = eptone: Configures the appropriate call progress tone for the local region. The call progress tone setting determines the dial tone, busy tone, and ringback tone to the ‘originating party . Configures a description for the voice port. You must use the description setting to describe the voice port in show command output. ILis always useful to provide some information about the usage of a port. The description ean specify the type of equipment that connects to the FXS port. . ring frequency (in Hz) for an FXS voice port, You ‘must select the ring frequency that matches the connected equipment. If itis set incorrectly, the attached telephone might not ring or might buzz. In addition, the ring frequeney is usually country-specific, You should take into account the appropriate ring frequeney for your area before you configure this command. a ring cadence: Configures the ring cadence for an FXS port. The ring cadence defines how ringing voltage is sent to signal a call. The normal ring cadence in North America is 2 seconds of ringing followed by 4 seconds of silence. In England, normal ring cadence is a short ring followed by a longer ring. When configured, the eptone setting automatically sets the ring cadence to match that country. You ean manually set the ring cadence if you want to override the default country value. You may have to shut down and reactivate the voice port before the configured value takes effect. = disconneet-ack: Configures an FXS voice port to remove line power if the equipment on an FXS loop-start trunk disconnects first. This removal of line power is not something the user hears. Instead, itis. method for electrical devices to signal that one side has ended the call = busyout: Configures the ability to busy out an analog port = station id name: Provides the station name associated with the voice port, This parameter is passed as a calling name (o the remote end if the call originated from this voice port. If no caller ID is received on an FXO voice port, this parameter is used as the calling name, The maximum string length is 15, = station id number: Provides the station number to use as the calling number associated with the voice port. This parameter is optional. When itis provided, it is wed as the calling number if the call originated from this voice port. [Fit is not specified, the calling number is determined from a reverse dial-peer search. Ifno caller ID is received on an FXO voice Port, this parameter is used as the calling number. The maximum string length is 15. a8 TP Telephony Express (PTX) v4.0 (© 2006 Cisco Systems, IncExample: FXS Voice Port Configuration Gq ‘Cutnowter (config) fvoice-port 4/070 ‘vor cuttRoster (config-voiceport) Heignal 1eop-start currouter (config-voiceport) feptone GB cutroster (config-vosceport) #ring cadence pattern02 cittRoster (config) #votce-port 1/0/1 cuprouter (contig-voiceport) #eignal Leop-start cutroster (contig-voiceport) fcptone GB cutkouter (config-voiceport) tring cadence pattern02 Example Revisit the scenario of an intemational company with offices in the United States and England. The figure shows how the British office is configured to enable loop-start signaling on a Cisco 2600 or 3600 Series Multiservice Platform on FSX voice port 1/0/0, The call progress tones are set for Great Britain, and the ring cadence is set for pattern 1 {© 2006 Cisco Systems, Ine. PSTN Inlerface and Voice Dial Peer Configuration 3-19Foreign Exchange Office Port Configuration FXO ports act like telephones and connect to central office (CO) switches or to a station port on a PBX. This topic identifies the configuration parameters that are specific to FXO ports FXO Voice Port Configuration Parameters signal ring number dial-type descripti supervisory disconnect Configuration Parameters In most instances, the FXO port connection functions with default settings. FXO port configuration allows you to set parameters based on the requirements of the connection when you need to alter default settings or parameters set for fine-tuning. You can set the following Configuration parameters '= signal: Sets the signaling type for the FXO port. Ifthe FXO port connects to the PSTN, the default settings are adequate. If the FXO port connects to a PBX, the signal setting must match the PBX = ring number: Configures the number of rings before an FXO port answers a call. This configuration is usefull when you have other equipment available on the line to answer incoming calls. The FXO port answers if the other equipment does not answer the incoming call within the configured number of rings. = dial-type: Configures the appropriate dial type for outbound dialing, Older PBXs or key ‘ets may not support dual tone multifrequency (DTMF) dialing, If you are connecting an FXO port to this type of device, you may need to set the dial type for pulse dialing = description: Configures a description for the voice port. Use the description setting to describe the voice port in show command output. 3.20 Tel sony Express (1PTX) v4.0 (© 2006 Cisco Systems, nc.Example supervisory disconnect: Configures supervisory disconnect signaling on the FXO port. Supervisory disconnect signaling is a power denial from the switch that lasts at least 350 ms. When this condition is detected, the system interprets it as a disconnect indication from the switch and clears the call, You should disable supervisory disconnect on the voice port if there is no supervisory disconnect available from the switch. Typically, supervisory disconnect is available when connecting to the PSTN and is enabled by default, When the connection extends out to a PBX, you should verify the documentation to ensure that supervisory disconnect is supported Example: FXO Voice Port Configuration FXO Port iro GvitRowter (config) Frais pore i/i/o cutRouter (config-votceport) feignal ground-start curouter (config-voiceport) #ring number 3 curouter (config-woiceport) #aial-type dime cottRouter (contig-votceport) #deseription Connection to_PS7H ‘The configuration in the figure enables ground-start signaling on a Cisco 2600 or 3600 Series Multiservice Platform on FXO voice port 1/1/0. The ring-number setting of 3 species that the FXO port does not answer the call until after the third ring. The dial type is set to DTMF. (© 2006 Ciseo Systems, ne PSTN interlace and Volos Dial Peer Configuration 21Ear and Mouth Port Configuration E&M ports provide signaling that you generally use for switch-to-switch or switch-to-network trunk connections. This topic identifies the configuration parameters that are specific to the E&M port E&M Voice Port Configuration Parameters + signal + operation + type + auto-cut-through + description 2 RiGee Configuration Parameters Although E&M ports have default parameters, you must usually configure these parameters to match the device that connects to the E&M port. You can set the following configuration parameters: signal: Configures the signal type for E&M ports and defines the signaling to use when notifying a port to send dialed digits, This setting must match that of the PBX to which the port is connected. You must shut down and reactivate the voice port before the configured Value takes effect. With wink-start signaling, the router listens on the M-lead to determine when the PBX wants to place a call. When the router detects current on the M-tead, it waits for availability of digit registers and then provides a short “wink” on the E-lead to signal the PBX to start sending digits, With delay-start signaling, the router provides current on the F-lead immediately upon seeing current on the M-lead. When current is stopped for the duration of the digit sending, the F-lead stays high until digit registers are available. With immediate-start signaling, the PBX simply waits a short time after raising the M-lead and then sends the digits without a signal from the router. operation: Configures the cabling scheme for E&M ports, The operation command affects, the voice path only. The signaling path is independent of two-wire versus four-wire settings. If you specify the wrong cable scheme, the user may get voive traffic in one direction only. You must match the settings of the device on the other end of the line. You must then shut down and reactivate the voice port for the new value to take effect. 3.22 IP Telephony Express (IPT) v8.0 (© 2006 Cisco Systems, ne= type: Configures the E&M interface type for a specific voice port. The type defines the electrical characteristics for the E-lead and the M-lead, The E-lead and the M-lead are monitored for on-hook and off-hook conditions. From a PBX perspective, when the PBX attempts to place a call, it goes high (off-hook) on the M-lead. The switch monitors the Mclead and recognizes the request for service. Ifthe switch attempts to pass a call to the PBX, the switch goes high on the E-lead. The PBX monitors the E-lead and recognizes the request for service by the switch. To ensure that the settings match, you must verify them with the PBX configuration. = auto-cut-through: Configures the ability to enable call completion when a PBX does not provide an M-lead response. For example, when the router is placing a call to the PBX, even though they may have the same correct signaling configured, the PBX may not provide the wink with the same duration or voltage. The router may not understand the PBX wink. The auto-cut-through command allows the router to send digits to the PBX, even when the expected wink is not detected, = description: Configures a description for the voice port. Use the description setting to describe the voice port in show command output Example: E&M Voice Port Configuration ‘isnouter config) Wolce-pare 1/370 cuerouter (config-voiceport)#aignal immediate ‘cutRouter (config-voiceport) #auto-cut-through cuttnouter (config-voicepert) Hoperation 4-wire ousrouter (config-voiceport) #type 1 cutRouter (config-voiceport) tes SETI Example ‘The configuration in the figure enables immediate signaling with automatic cut-through for an E&M connection to an MOH device, This configuration allows an external device to provide ‘MOH to the Cisco Unified CallManager Express system. The type setting matches the E&M port setting on the MOH deviee as well as the number of wires used by the operation command. (© 2006 Cisco Systems, nc ‘PSTN interface and Voloe Dial Peer Configuraton 3.23,Timers and Timing This topic identifies the timing requirements and adjustments that are applicable to voice interfaces. Under normal use, these timers do not need adjusting. When ports connect to a device that does not properly respond to dialed digits or hookflash or when the connected device provides automated dialing, you can configure these timers to allow more or ess time for a specific function Timers and Timing Configuration be Ce + timeouts + timeouts interdigit + timeouts ringing + timing digit + timing interdigi + timing hookflash-in and hookflash-out Configuration Parameters You can set a number of timers and timing parameters to fine-tune the voice port, Following are voice port configuration parameters that you can set neouts initial: Configures the initial digit timeout value in seconds. This value controls, how long the dial tone is presented before the first digit is expected. This timer typically does not need to be changed, = timeouts interdigit: Configures the number of seconds that the system waits for the next digit after the caller has input the initial digit. If the digits are coming from an automated device and the dial plan is a variable-length dial plan, you can shorten this timer so that the call proceeds without having to wait the full default of 10 seconds for the interdigit timer to expire = timeouts ringing: Configures the length of time that @ caller can continue ringing a telephone when there is no answer. You can configure this setting to be less than the default of 180 seconds so that you do not tie up the voice port when it is evident that the call is not going to be answered. if digit: Configures the DIMF digit-signal duration for a specified voice port. You can use this setting to fine-tune a conneetion to a device that may have trouble recognizing dialed digits. Ifa user or device dials too quickly, the digit may not be recognized. By changing the timing on the digit timer, you can provide a shorter or longer DTMF duration, 324 IP Telephony Express (PTX) v4.0 {© 2006 Cisco Systems, nc= timing interdigit: Configures the DIMF interdigit duration for a specified voice port. You can change this setting to accommodate faster or slower dialing characteristics. m= timing hookflash-in and hookflash-out: Configures the maximum duration (in milliseconds) of a hookflash indication. Hookflash is an indication by a caller that the caller wishes to do something specific with the call, such as transfer the call or place the eall on hold. For hookflash-in, the FXS interface processes the indication as on-hook if the hhookflash lasts longer than the specified limit. If you set the value too low. the hookflash ‘may be interpreted as a hang-up. If you set the value too high, the handset has to be left hhung up for a longer period to clear the call. For hookflash-out, the setting specifies the duration (in milliseconds) of the hookflash indication that the gateway generates outbound. ‘You can configure this value to match the requirements of the connected device. Example: Timers and Timing Configuration GuBRowter (config) Fvoice-pore 1/070 curRouter (config-vasceport) Nasgnal Leop-start comnouter (config-voiceport) feptone GB cupnouter (contig-voiceport) #ring cadence pattern0i cumnouter (contig-voiceport) #eimecuts initial 15 curouter (config-votceport) #einecuts Interdigit 15 currouter (config-voiceport) #timeouts ringing 120 crnouter (config-voiceport) #tining heok#lash-in 500 Example The installation in the figure is fora facility for elderly residents, Users in such a facility may need more time to dial digits than is typical. They may also want the telephone to ring unanswered for only two minutes, The configuration in the figure enables several timing parameters on a Cisco voice-enabled router voice port 1/0/0. The initial timeout is lengthened to 15 seconds, the interdigit timeout is lengthened to 15 seconds, and the hookflash-in timer is set t0 500 ms, {© 2008 Cisco Systems, In. PSTN Inierface and Voice Dial Peer Configuration 3-25Digital Voice Port Configuration ‘This topic identifies the configuration parameters that are specific to TI and El digital voice ports. Basic T1 or E1 Controller Configuration CRCA, framing SF, ESF No-CRC4, Australia linecode ‘AMI, B8ZS. ‘AMI, HDB3 clock source | Line, internal | Line, internal n Parameters When you purchase a 1 or EI connection, make sure that your service provider gives you the appropriate settings. Before you configure a TI or El controller to support digital voice ports, ‘you must enter the following basic configuration parameters to bring up the interface: framing: Selects the frame type for a Tl or EI data line. The framing configuration differs, between TI and EI Options for T' — — Options for E — Default for TI: SF Default for El: CRC4 inecode: Configures the line-encoding format for the digital service level 1 (DS1) link, juper Frame (SF) or Extended Super Frame (ESF) clic redundancy check 4 (CRC4), no-CRC4, or Australia Options for T1: Alternate mark inversion (AMI) or binary 8-zero substitution (B87S) Options for E1: AMI or high-density binary 3 (HDB3) Default for T1: AMI Default for E1: HDB3 clock source: Configures clocking for individual T1 or El links. Options: Line or internal Default: Line 3.28 IP Telephony Express (PTX) v4.0 (© 2008 Cisco Systems, Inc.Basic T1 or E1 Controller Configuration (Cont.) cuRouter (contig-control) # linecode {ami | béze) * Configures the line code for a T1 line teRouter (config-contzol) # Linecods {ami | haba} + Configures the line code for an E1 line Use the linecode command to identify the physical layer signaling method to satisfy the ones density requirement on the digital facility of the provider. Without a sufficient number of ones in the digital bitstream, the switches and multiplexers in a WAN can lose theit synchronization for transmitting signals. The table shows the linecode command. linecode Command Command Description ami Alternate mark inversion; used for T1 configurations baze Binary 8-20r0 substitution; used for T1 PRI configurations hab3 High-density binary 3, used for Et PRI configurations B8ZS accommodates the ones density requirements for TI carrier facilities using special binary signals encoded over the digital transmission link. It allows 64 kbps (clear channel) for ISDN channels. The settings for these two Cisco IOS software controller commands on the router must match the framing and line-code types used at the TI/EL WAN CO switch of the provider. TI configurations typically require the framing esf command and the linecode b8zs command. EI configurations typically require the linecode hdb3 command (© 2006 Cisco Systems, ine. ‘PSTN interface and Voce Dial Peer Configuration 3.27Basic T1 or E1 Controller Configuration (eens) . : coutnovter(config-control)# traning eat} + Configures the framing for aT1 line corouter (config-controt) # ‘framing (ered | no-cxed | australia) + Configures the framing for an E1 line eee Use the framing command to select the frame type used by the PRI service provider. The table shows framing controller configuration commands that you can use. framing Command Command Description af ‘Super Frame; used for some older T1 configurations f Extended Super Frame; used for T1 PRI configurations ered or no-crcd Cyclic redundancy check 4; used for E1 PRI configurations. Note ESF and CRC4 are most common in new Ts or Ets, 3:28 IP Telephony Express (IPTX) v4.0 (© 2006 Cisoo Systems, ne.Channel Associated Signaling Configuration This topic describes the commands required to configure a CAS interface Basic T1 or £1 Controller Configuration c(erelia} CusRouter (contig-control) # ‘ds0-group da0-group-no timeslots timeslot-list type {exm-delay-dial | eam-fgd | ckm-inmediate-start | esm-wink-etart | ext-sig | fgd-eana | fxo-ground- start | fxo-loop-start | fxs-ground-start | fxs- loop-start} + Creates the voice ports of the T1 or E1 and the signaling that is used curnouter (contig-control) # clock source {line | internal} + Sets the source of the clocking ‘You must ereate a digital voice port in the TI or El controller to make the digital voice port available for specific voice port configuration parameters. You must also assign time slots and signaling to the logical voice port through configuration. The first step is to create the TI or EL digital voice port with the ds0-group ds0-group-no timeslots rimeslot-lst type signal-type command ‘The ds0-group keyword automatically creates a logical voice port that is numbered as ds0- group-no. The dS0-group-no parameter identifies the digital service level 0 (DSO) group (numbered from 0 to 23 for TI and from 0 to 30 for E1). This group number is used as part of the logical voice port numbering scheme. The timeslots keyword allows you to specify which time slots are parts of the DSO group. The simestor-list parameter is a single time-slot number, a single range of numbers, or multiple ranges of numbers separated by commas. ‘The type keyword defines the emulated analog signaling method that the router uses to connect to the PBX or PSTN. The type depends on whether the interface is TI or BI ‘To delete a DSO group, you must first shut down the logical voice port, When the port is in a shutdown state, you can remove the DS0 group from the TI or El controller with the no ds0-group ds0-group-no command. Use the clock source {line | internal} command to configure the 'T! and El clock source on Cisco routers. (© 2006 Ciseo Systems, ne PSTN Interface and Voice Dial Peer Configuration 3-29Example: Basic T1 or E1 Controller Configuration XS ‘BébRowterIeontighecontrotier FH I/E eenouterlcontig-cont roller) tclock sour ‘eenouter[eontig-conteolier)#Linecode habs a = pr er ‘The first example configures the TI controller for ESF, B8ZS line code, and time slots 1 through 24 with FXO ground-start signaling, The resulting logical voice port is 1/0:1, where ~1/0" is the module and slot number and ":1” is the ds0-group-no value that was assigned during configuration, The El configuration uses a line code of HDB3, framing of CRC4, and time slots of I through 15 with E&M wink-start signaling. The resulting logical voice port is 1/0:1, where “1/0” is the ‘module and slot number and”:1” is the ds0-group-no value that was assigned during the configuration. 3:30 IP Telephony Express (PTX) v4.0 (© 2006 Cisco Systoms, nc.Common-Channel Signaling: BRI This topic identifies the most common components and reference points of ISDN BRI, and it provides an overview of the configuration commands required to successfully configure an ISDN BRI connection, including an over command. And finally, because you may have to configure the Layer 2 bearer (B) channel encapsulation protocol and authentication when configuring ISDN BRI, this topic shows you how to do that BRI Reference Points To Non-ISDN Device (TE2) To ISDN Late Service cisco tson AE, ST __, uv Router NTT a (Tet) 4.Wire 2Wire Circuit ci it fal “wy irc ireut_ Wall, ISDN Phone (TE) ed There are many ISDN interface abbreviations, such as T, S, U, S/T, and so on. What do all of these components and reference points look like in practice” When creating a network, connect the Network Termination 1 (NT-1) to the wall jack with a standard two-wire connector, then to the ISDN phone, terminal adapter, Cisco ISDN router, and maybe a fax with a four-wire connector. The S/T interface uses an eight-wire connector (two pairs for data transmission and two pairs for providing optional power to the network terminal [NT] and the terminal endpoint (TE). Take care when connecting ISDN devices, because RJ-I and RJ-4S connectors look similar. The S/T reference point is as follows: = Four-wire interface = Point-to-point and multipoint (passive bus), as shown in the figure = Covered by the ITU-T 1.430 physical layer specification for BRI interfaces and the ANSI TI.601 standard for the United States ‘The SIT interface defines the interface between a TE1 or terminal adapter and a network terminal. A maximum of eight devices can be daisy-chained to the S/T bus. (© 2006 Cisco Systems, nc. PSTN Interface and Voice Dial Peer Configuration 3.31‘The U interface defines the two-wire interface between the NT-I and the ISDN cloud. The United States uses the U interface; the rest of the world uses an S/T interface ‘The R interface defines the interface between the terminal adapter and an attached non-ISDN device (TE2), In North America, the NT-1 function is commonly integrated into the ISDN device (router, terminal adapter), thus permitting a direct connection from the ISDN device to the telco jack. An NT-1 and NT-2 combination device is sometimes referred to as an NTU. In most countries, the service provider (telco) provides the NT-I and NT-2 combination, and customer access is available only at the S/T interface. ISDN Configuration Tasks. h type, either globally or on an interface. + The interface setting overrides the global setting. * Configure the interface or controller settings. To configure an ISDN BRI interface on a router, you must specify global and interfuce configuration commands. = Global configuration tasks include these: Select the switeh type that matches the ISDN provider switch at the CO. — Set destination details. Indicate static routes from the router to other ISDN destinations. Specify the traffic criteria that initiate an ISDN call to the appropriate destination = Interface configuration tasks include these: — Select the ISDN BRI port and configure an IP address and subnet mask. ~ Although the interface automatically inherits the global switch type setting, some configurations may require you to configure a specific switch type on an interface. — Configure optional features, including length of time for the ISDN carrier to wait before responding to the call and seconds of idle time before the router times out and drops the call 3.32 IP Telephony Express (IPTX) vt 0 (© 2006 Ciseo Systems. ne.ISDN BRI Configuration Commands currouter (config) # isdn ewitch-type aviteh-type + Sots the ISDN switch type globally (contig) + interface BRI nod/port + Designates the interface used for ISDN on a router acting as a TE! device curRouter (config-1#)4 isdn switch-type switch-cype + Sets the ISDN switch type on an interface {overrides the global setting if it exists) ERS At the global level, the administrator must specify the ISDN service provider CO switch type. There are several types of switches to choose from, and some of these require special parameters. The specifics of standards signaling differ by region. Therefore, the switch type varies according to its geographical location. For example, the DMS-100 and National ISDN-1 type require you to specify a service profile identifier (SPID). This setting is optional on some switches (for example, the AT&T SESS) or not required at all ‘The interface bri inrerface-number command designates the interface used for ISDN on a router acting as @ TE! device, A router without a native BRI interface is a TE2 device. It must connect to an external ISDN terminal adapter via a serial interface. On a TE2 router, you must use the interface serial interface-rumber command, 1© 2006 C800 Systems, ne PSTN interface and Voice Dial Peer Configuration 3.33Use the isdn switch-type command to specify the CO switch to which the router connects. This table shows the possible switch types and their corresponding commands for BRI ISDN. isdn switch-type Commands Command | basic-5ess basic-dmsi00 Description [ATAT basic rate switches (United States) NT DMS-100 (North America) basic-ni basic-asig basic-net3 PINX (PBX) switches with QSIG signaling per Q.931 ‘National ISON-1 (North America) NETS switch type for United Kingdom, Europe, Asia, and Australia No switch defined Note Other switch types are available, The list of switch types can differ based on the Cisco 10S software version used ‘When you use the isdn switch-type command in global configuration mode, all ISDN interfaces on the router are configured for that switch type. Beginning with Cisco IOS Software Release 11.3T, the isdn switeh-type interface configuration mode command was introduced to allow different interfaces to be configured with different switch types. If you use the command in interface configuration mode, only the interface that is configured assumes that switch type. The interface setting always overrides the global setting 334 IP Telephony Express (PTX) v4.0 {© 2006 Cisco Systems, In.ISDN BRI Configuration Commands (Cont.) cutrouter (config-int) # iedn spidl spid-number Uda + Defines SPID 1 if assigned by the carrier (found in North America) comnouter (config-int) dedn spid? spid-number (ida) + Defines SPID 2 if assigned by the carrier (found in North America) Bit ceca see Several ISDN service providers use CO switches that require SPIDs. SPIDs authenticate call requests that are within contract specifications. These switches include National ISDN and DMS-100 ISDN switches, as well as the AT&T SESS multipoint switch. SPIDs are used only in the United States and are typically not required for ISDN data communications applications. The service provider supplies the local SPID numbers. If you are uncertain, contact the service provider to determine whether you need to configure SPIDs on your access routers. Use the isdn spidI and isdn spid2 commands to access the ISDN network when your router ‘makes its call to the local ISDN exchange. The table shows the isdn spid1 command syntax for the first BRI 64-kbps channel isdn spid1 and isdn spid2 Commands ‘Commands Description spid-number ‘Number identifying the service to which you have subscribed. This value is usually a 10-digt telephone rhumper followed by more digits. The ISDN service provider assigns this value lan (Optional) Seven-digit iocal directory number that is assigned by the ISDN service provider. Ifyou want to automatically detect the SPID, you can specify 0 for the spid-mumber argument. ‘The dn parameter allows you to associate up to three local directory numbers with each SPID. This number must match the called-party information coming in from the ISDN switeh in order for both B channels to be used on most switches. (© 2005 Cisco Systems, Ine. PSTN Interface and Voice Dial Peer Configuration 3.35ISDN BRI Configuration Example BRI OM CaERouter (config) ¥ieds ewites-type basie-Se cumnouter (contig) #interface bri 0/2 cumrouter (contig-int)#isdn pid 20655512340001, eutrouter (config-int) Aide apia? 2065551234002 This figure shows an example of configuring a BRI interface. The ISDN switch type is set to basic-Sess, and SPIDs of 2065551234001 and 2065551234002 have been configured, 3.86 IP Telephony Express (PTX) vl 0 (© 2006 Cisco Systems, ine.Common-Channel Signaling: PRI This topic identifies the most common components and reference points of ISDN PRI. It also shows how to use global and interface configuration commands to configure ISDN PRI and provides an overview of the isdn switeh-type command. In addition, the topic lists and explains the commands required to configure the ISDN PRI channels and data (D) channel, bere mela ts | csur | we T DSU T + ITU-T 1.431 + ANSIT1.601 + ITU-T 1.430 Depending on the country implementation, either the ANSI T1.601 or ITU-T 1.431 standard ‘governs the physical layer of the PRI interface. PRI technology is a bit simpler than BRI technology. The wiring is not multipoint, which refers, to the ability to have multiple ISDN devices connected to the network, all of which have access to the ISDN network. Arbitration at Layer | and Layer 2 allows multiple devices that need to share the ISDN network to access the network without collisions or interruptions. But because there are no multiple devices in PRI, it does not require this arbitration. There is only the straight connection between the CSU/DSU and the PRI interface. ‘© 2006 Cisco Systems, Ine. PSTN interface and Volee Dal Peer Configuration 3-37ISDN PRI Configuration Commands cuprouter (contig) # Leda owitch-type switch-type + Sets the ISDN switch type globally cutRouter (contig) controller {ti | el) {elot/port | unit-number} configuration mode cutrouter (config-iE)4 * Configures a controller as either T1 or E1 and enters controller isdn switch-type switch-type + Sets the ISDN switch type on an interface (overrides the global setting if it exists) Use the isdn switeh-type command to specify the CO PRI switch to which the router connects, With Cisco IOS Software Release 11.3(3)T or later, this command is also available as an interface command to allow different interfaces to support different switches. If you configure the switch type as a global command, the switch type applies to all interfaces uniess you specifically configure a switch type on an interface. An incompatible switch selection configuration can result in failure to make ISDN calls, After changing the switch type, you must reload the router to make the new configuration effective. This table shows the available options for the isdn switeh-type command. isdn switch-type Command ‘Command Description primary-éess primary-Sess “primary-dns100 primary-ni “primary-nte primary-nets primary-qsig Unlike BRI operation, configure SPIDs, regardl ATT primary 4€SS switches (United States) ATAT primary SESS switches (United States) NT DMS-100 switches (North America) National ISON switch type NTTISDN PRI switches (Japan) European and Australian ISON PRI switches QSIG signaling per 0.931 [ No switch defined DN PRIs do not use SPIDs. Therefore, there is no requirement to 8 of the ISDN switch type used by the PRI 338) IP Telephony Express (PTX) v4.0 (© 2006 Cisco Systems, neUse the controller {t1 | e1} slot/port command in global configuration mode to identify the controller to be configured. Use a single unit-mumber value to identity the Cisco ASS000, Series Universal Gateway controller. controller {t1 | e1} Command Command Description 2 ‘Specifies the controller interface for North America and Japan et ‘Specifies the controler interface for Europe and most other counties inthe world slot/port or unit- | Specifies the physical slotiport location or unit number of the number controller (© 2006 Cisco Systems, ne PSTN Interface and Voice Dal Peer Configuration 339ISDN PRI Configuration Commands (Cont.) ‘cusrouter (config-controller) PFi-group timeslots range + Sets the PRI group with a range of time slots cusroute: (contig-int)# Anterface serial elot/port: timeslot + Sets the PRID channel erro ‘The pri-group command configures the specified interface for PRI operation and specifies which fixed time slots (channels) are allocated on the digital facility of the provider: pri-group Command ee Command | Description timeslots range ‘The range of time siots allocated to this PRI. For T1, use. values in the range of 1 to 24, and for E1, use values {rom 1 to 31. The speed of the PRI s the aggregate of the channels assigned Ifusing all 30 B channels on an El PRI (30 B + D), specify pri-group 1-31 Ifonly the first eight B channels (512 kbps total data bandwidth) are allocated for a TI PRI (23 B + D), then specify pri-group 1-8,24. Note that you must specify the D channel. Note When provisioning a PRI line with fewer than 24 time slots (or 30 for E1), include the D ‘channel for signaling, 340 IP Telephony Express (1PTH) v8.0 {© 2006 Cisco Systems, In.Specification of the PRI group automatically creates the corresponding serial interface for the D channel; terface serial {slot/port | unit}:{23 | 18}. You use this interface to configure the PRID channel. The table shows the interface serial command options that you can use. interface serial Command ‘Command Description slot/port | The slotpor of the channelized controller uni The unit number of the channelized controller on a Cisco 4000, or | A89000 Series Router 23 ‘ATT interface that designates channelized DS0s 0 to 22 as the B ‘channels and O80 23 as the D channel 15 ‘An E1 interface that designates 30 B channels and time siot 16 as, the D channel Note Inan Et or T1 facility, the channels start numbering at 1 (1 to 31 for E1 and 1 to 24 for T1), Serial interfaces in the Cisco router start numbering at 0. Therefore, channel 16, the Et signaling channel, is serial port subinterface 18. Channel 24, the T1 signaling channel, is sorial subintorface 23. {© 2006 Cisco Systems, ne. PSTN Interface and Voice Dial Peer Configuration 3-41,Example: ISDN PRI Configuration PRIO/ ‘CMERouter (config) #isdn ewitch-type primary-ni cMmRouter (config) #controller ti 0/1 cMBRouter (config-controller) #pri-group timeslots 1-24 cMERouter (config-controller) #framing esf cMBRouter (config-controller) #linecode bez cMBRouter (config-controller) #eleck source line CMERouter (config) #interface serial 0/0:23 The table describes the ‘ommands found in the figure. PRI Configuration Commands Command [ beseription Tedn ewitch-type Selets a swich ype of National SON peimary-at controller ti 0/1 Selects the TH controler pri-group timeslots 1-24 | Establishes the interface port to function as PRI with 24 time slats (including D channel) designated to operate at a speed (of 64 kbps framing esf | Selects ESF framing, a T1 configuration feature linecode b8zs | Selects line code B8ZS for T1 clock source line ‘Specifies the T1 line as the clock source for the router interface serial 0/0:23 | Identiies the D channel on serial interface 010 The controller t1 0/1 command configures the TI controller. In the example, the switch type that is selected is the National ISDN standard. This example is accurate for some operations in the United States. For an E1 example, the time slot argument for the pri-group command would be I-31 rather than 1-24, as shown for the TI example, and the interface command would be 0/1215 instead of O/1:23, 3:42 __IP Telephony Express (PTX) v8.0 (© 2006 Cisco Systems, ne.Summary ‘This topic summarizes the key points that were discussed in this lesson Sela + Configurable parameters on FXS ports include signal, cptone, description, ring frequency, ring cadence, disconnect-ack, busyout, station id name, and station id number. Configurable parameters on FXO ports include signal, ring number, dial-type, description, and supervisory disconnect. Configurable parameters on E&M ports include signal, operation, type, auto-cut-through, and description. + Configurable timer and timing parameters define initial digit and interdigit timing, digit and interdigit duration, as well as ringing time. + You must configure the following parameters on T1 and E1 lines before they can support digital voice: framing, linecode, and clock source. Summary (Cont.) + Digital voice ports are created with the ds0-group command in the TH/E1 controller. + Enabling ISDN BRI requires global configuration and interface configuration commands. + ISDN PRI configuration requires that the pri-group command specify the time slots that are used for voice and signaling. ISDN PRI does not require SPIDs. Sam SER (© 2005 Cisco Systems, Ine PSTN interface and Voice Dial Peer Configuration 3.43Lesson 3 Configuring Dial Peers Overview ‘This lesson describes VoIP dial peers, plain old telephone service (POTS) dial peers, how the system matches dial peers, and dial peer 0 Objectives Upon completing this lesson, you will be able to describe dial peers and configuration tasks. This ability includes being abie to meet these objectives: © Describe dial peers and their application = Configure POTS dial peers © Configure VoIP dial peers = Describe destination pattern options and the applicable shorteuts = Describe how inbound dial peers are matched = Describe dial peer 0 Describe how outbound dial peers are matchedWhat Is a Dial Peer? This topic describes dial peers and their applications. Raut vee MPI ELe aad + Adial peer is an addressable call endpoint. peers establish logical connections, or call legs, to complete an end-to-end call + Cisco voice-enabled routers support two types of dial peers: — POTS dial peers: Connect to a traditional telephony network like FXO, FXS, E&M, BRI, PRI, T1, £1 - VoIP dial peers: Connect over a packet network using an IP address. When a call is placed, an edge device generates dialed digits as a way of signaling where the call should terminate. When these digits enter a router voice port, the router must have a way to decide whether it can route the call and where the call can be sent. The router does this by looking through a list of dial peers A dial peer isan addressable call endpoint. The address is ealled a destination partern and is configured in every dial peer. Destination patterns can point to one telephone number only or to 4 range of telephone numbers. Destination patterns use both explicit digits and wildcard variables to define a telephone number or range of numbers. ‘The router uses dial peers to establish logical connections. The router establishes these logical connections, known as call legs, in either an inbound or outbound direction. Dial peers define the parameters for the calls that they match, For example, ifa eall originates and terminates at the same site and does not eross through slow-speed WAN links, then the call, can cross the local network uncompressed and without special priority. A call that originates locally and crosses the WAN link to a remote site may require compression with a specifi codec. In addition, this call may require that voice activity detection (VAD) be turned on, and it will need to receive preferential treatment by being given a higher priority level 38.48 IP Telophony Express (PTX) vt 0 {© 2006 Cisco Systems, IneCisco Systems voice-enabled routers support two types of dial peers: = POTS dial peers: Connect to a traditional telephony network, such as the public switched telephone network (PSTN) or a PBX, or to a telephony edge device, such as a telephone or fax machine. POTS dial peers perform these functions: Provide an address (telephone number or range of numbers) for the edge network or device — Point to the specific voice port that connects the edge network or device © VoIP dial peers: Connect over a packet network. VoIP dial peers perform these functions: — Provide a destination address (telephone number or range of numbers) for the edge device that is located across the network ociate the destination address with the next-hop router or destination router, depending on the technology used {© 2006 Cisco Systems, nc PSTN interlace and Voice Dial Peer Configuration 3-47ye Voice-Enabled Telephony Device Voice-Enabled Packet In the figure, the telephony device connects to the Cisco Systems voice-enabled router POTS dial peer. The POTS dial peer configuration includes the telephone number of the telephony device and the voice port to which it attaches. The router knows where to forward incoming calls for that telephone number. The Cisco voice-enabled router VoIP dial peer connects to the packet network, The VoIP dial peer configuration includes the destination telephone number (or range of numbers) and the next-hop or destination voice-enabled router network address. Follow the steps in the table to place a VoIP call How to Place a VoIP Calll Stop Action 1 Configure the source router with a compatible dial peer that species the recipient destination address. 2 Configure the recipient router with a POTS cial peer that specifies which voice Port the router uses to forward the voice call 348 IP Telephony Expeass (PTX) v4.0 {© 2006 Cisco Systems, nc.Plain Old Telephone Service Dial Peers This topic describes how to configure POTS dial peers. POTS Dial Peers 1234 4aO™ FXS 1/014 (ainoutar (confighWalal-poor voice 20 pot cumnouter (contig-dialpeer) #destination-pattern 1234 OneRouter (contigrdialpear) ¢port 1/0/3 eM ahaa See Before you can begin the configuration of Cisco IOS dial peers, you must have a good understanding of where the edge devices reside, what type of connections need to be made between these devices, and what telephone numbering scheme is applied to the devices. Follow the steps in the table to configure POTS dial peers. How to Configure POTS Dial Peers Step ‘Action 1 Configure a POTS dial peer at each router or gateway where edge telephony devices connect to the network 2 Use the destination-pattern command in the dial peer to configure the telephone number. 3 Use the port command to specify the physical voice port to which the POTS telephone connects, You specify the dial peer type as POTS because the edge device directly connects to a voice port and the signaling must be sent from this port to reach the device. The two basic parameters that you must specify for the device are the telephone number and the voice port. When 2 PBX is connecting to the voice port, you can specify a range of telephone numbers. {© 2006 Cisco Systems, Inc PSTN Interface and Voice Dial Peer Configuration S49Example The figure illustrates proper POTS dial peer configuration on a Cisco voice-enabled router. The dial-peer voice 20 pots command notifies the router that dial peer 20 is a POTS dial peer with a tag of 20. The destination-pattern 1234 command notifies the router that the attached telephony device terminates calls destined for telephone number 1234, The port 1/0/1 command notifies the router that the telephony device is plugged into module I, voice interface card (VIC) slot 0, voice port 1 3:50 IP Telephony Express (PTX) vl.0 {© 2006 Ciseo Systems, ne.VoIP Dial Peers ‘This topic describes how to configure VoIP dial peers. VoIP Dial Peers wicca] [waar Stan | Phone 1734 comERoutert cueRouter2 2010 aa A _—. fate ge so10101 ro20.102 GERouteri (config) #dial-peer voice 20 pots cueRouteri (config-dialpeer) #destination-pattern 1234 CMERoUter (config-dialpeer) Hport 1/0/21 cueRouter: (config) #dial-p 1 voice 30 voip cupRouter! (config-dialpeer) #destination-pattern 2. cMERouterl (config-dialp don target {pv4:10.10.10.2 The administrator must know how to identify the far-end voice-cnabled device that will terminate the call. In a small network environment, the device may be the IP address of the remote device. In a large environment, the device may mean pointing to another router or gatekeeper for address resolution and Call Admission Control (CAC) to complete the call. You must follow the steps in the table to configure VoIP dial peers. How to Configure VoIP Dial Peers Step Action 1 Configure the path across the network for voice data 2 ‘Speaty the dial peer as a VoIP cal peer. 3 - Use the destination-pattern command to configure a range of numbers reachable by the remote router or gateway. 4 Use the session target command to specily an IP address of the terminating router or gateway. 5 Use the remote device loopback address as the IP address. ‘You specify the dial peer as a VoIP dial peer, which alerts the router that it must process a call according to the various parameters that you specify in the dial peer. The dial peer must then package the voice traffic as an IP packet for transport across the network, Specified parameters ‘may include the codec to be used, whether to use Real-Time Transport Protocol (RTP) header compression, and whether to use VAD and may include marking the packet for priority service © 2006 Cisco Systems, Ine. PSTN Interface and Voce Dial Peer Configuration 3-81‘The destination-pattern parameter configured for this dial peer is typically a range of numbers that are reachable via the remote router or gateway. Because this dial peer points to a device across the network, the router needs a destination IP address to put inthe IP packet. The session target parameter allows the administrator to specify an IP address either of the terminating router or gateway or another device; for example, a gatekeeper that can return an IP address of that remote terminating device It is recommended that you use a loopback address to determine which IP address a dial peer should point to, The loopback address is always up on a router as long as the router is powered (on and the interface is not administratively shut down. If you use an interface IP address instead of the loopback and that interface goes down, the call fails even if there is an alternate path to the router, Example ‘The figure illustrates a typical dial peer configuration on a Cisco voice-enabled router. The dial-peer voice 20 pots command notifies the router that dial peer 20 is a POTS dial peer with 4 tag of 20. This dial peer is matched in the inbound direction The dial-peer voice 30 voip command notifies the router that dial pecr 30 is a VoIP dial peer with a tag of 30. The destination-pattern 2... command notifies the router that this dial peer defines an IP voice path across the network for telephone numbers 2000-2999. The session target ipv4:10.10.10.2 command defines the IP address of the router that is connected to the remote telephony device. Dial peer 30 is used in the outbound direction, VoIP Dial Peers (Cont.) (nae na aaa vor sox0101 tox0102 crerzouter2(contig-dtalpoer) #dectination-pattern 1236 ueexouter2(contig-dtalpeer) port 2/0/1 urnouter2 (config-dialpeer) teesoion target spv4+i0.10,10.2 Example ‘The figure illustrates the proper VoIP dial peer configuration on a Cisco voice-enabled router. The dial-peer voice 40 voip command notifies the router that dial peer 40 is a VoIP dial peer with a tag of 40 This dial peer is matched in the inbound direction. In the outbound direction, an Ethernet phone directory number (ephone-din) is defined and is matched, 352 IP Telephony Express (IPT) v8.0 '© 2006 Gis00 Systems, ne.Destination Pattern Options This topic describes destination pattern options and the applicable shortcuts. bey url Me elute) life Ed Common destination pattern wildcards + Period () Specifies any one wildcard digit + Comma (,) ~ Inserts a one-second pause + Square brackets Indicates a range of digits within the brackets + Plus (+) ~ The preceding digit occurred one or more times. + Percentage (%) The proceding digit occurred zero or more times oT ~ Indicates a variable-length pattern + Asterisk (*) and pound sign (#) = Not valid wildcards; are DTMF tones ‘The destination pattern associates a telephone number with a given dial peer. The destination pattern also determines the dialed digits that the router collects and forwards to the remote telephony interface, such as a PBX, Cisco Unified CallManager, Cisco Unified CallManager Express router, Cisco IOS router, or the PSTN. You must configure a destination pattern for each POTS and VoIP dial peer that you define on the router, The destination pattern can indicate a complete telephone number of a partial telephone number with wildcard digits; it can also point to @ range of numbers defined in a variety of ways. Destination pattern options include these: = Plus (+): An optional character that indicates an E.164 standard number. E.164 is the ITU-T recommendation for the international public telecommunication numbering plan. The plus sign in front ofa destination patter string specifies that the string must conform, to Recommendation E.164. m= String: A series of digits specifying the E.164 or private dialing-plan telephone number. The following examples show the use of special characters that are often found in destination patterns strings Asterisk (*) and pound sign (#) appear on standard touch-tone dial pads. You may need to use these characters when passing a call to an automated application that requires these characters to signal the use of a special feature. For example, when calling an interactive voice response (IVR) system that requires a code for access, the number dialed might be "5551212888#", which would initially dial the telephone number 5551212 and input a code of 888 followed by the pound key to terminate the IVR input query {© 2006 Ciseo Systems, nc PSTN Interface and Vows Dial Per Configuration 359,— Comma (,) inserts a one-second pause between digits. For example, you can use the ‘comma where a9 is dialed to signal a PBX that the call should be processed by the PSTN. The 9 is followed by a comma to give the PBX time to open a call path to the PSTN, after which the remaining digits are played out. An example of this string is 9,5351212. — Period (.) matches any single entered digit (use this character as a wildcard). Use the wildcard to specify a group of numbers that may be accessible via a single destination router, gateway, PBX or Cisco Unified CallManager Express router. Because the period (commonly referred to as a dot) indicates a single digit of 0 t0 9, this pattern limits how efficiently ranges of numbers are used. A pattern of “200.” allows for 10 uniquely addressed devices, whereas a pattern of “20..” can point to 100 devices. If one site has the numbers 2000 through 2049 and another site has the ‘numbers 2050 through 2099, then the bracket notation would be more efficient — Brackets ([]) indicate a range. A range isa sequence of characters that are enclosed in the brackets. Only single numeric characters from 0 to 9 are allowed in the range In the previous example, the bracket notation could be used to specify exactly which range of numbers is accessible through each dial peer. For example, the first site pattern would be “20[0-4},” and the second site pattern would be “20[5-9].” The bracket notation offers much more flexibility in how you can assign numbers. = T: An optional control character indicating that the destination pattern value is a variable-length dial string. In cases where callers may be dialing local, national, or international numbers, the destination pattern must provide for a variable-length dial plan. Ifa particular voice gateway has access to the PSTN for local calls and access to a transatlantic connection for international calls, then calls being routed to that gateway will have a varying number of dialed digits. A single dial peer with a destination pattern of *.T" could support the different call types. The interdigit timeout determines when a string of dialed digits is complete. The router continues to collect digits until there is an interdigit pause longer than the configured value, which by default is 10 seconds. When the calling party finishes entering dialed digits there is a pause equal to the interdigit timeout value before the router processes the call. The calling party can immediately terminate the interdigit timeout by entering the pound (#) character, which is the default termination character. Because the default interdigit timeout is 10 seconds, users may experience a long call setup delay, Note Cisco 10S software does not check the validity of the E. 164 telephone number; it accepts any series of digits as a valid number. IP Telephony Express (PTX) v4.0 (© 2006 Cisoo Systems neExample Example: Destination Pattern Options Destination Pattern Matching Telephone Numbers 9551234 5551-3] Matches one telephone number exactly, 8551234, This destination pattem is typically used when there isa single device, such asa telephone or fax, connected to a voice part. the fourth digt can be 1, 2, or 3, and the last digits can be any valic digits This type of destination pattern is used when telephone number ranges are assigned to specific sites. in this example, the destination pattern is used in 3 ‘small site that does not need more than 30 numbers assigned, ‘Matches any telephone number that has at least one digit and can vary in length from 1 to 32 digits total, ‘This destination pattern is used fora dial peer that services a variable-length dial plan, such as local, national, and intemational calls. it can also be used 238 a default destination pattern so that any calls that do not match a more specific pattern wil match this one and can be directed to an operator. ‘Matches a seven-cigit telephone number where the first three digits are 555, {© 2006 Cisco Systems, ine. PSTN interface and Voice Dial Peer Configuration 355Matching Inbound Dial Peers ‘This topic describes how the router matches inbound dial peers. Matching Inbound Dial Peers Inbound matching behaviors: + incoming called-number Defines the called number or Dialed Number Identification ice (ONIS) string + answer-address: Defines the originating calling number or Automatic Number Identification (ANI) string + destination-pattemn Uses the calling number (originating or ANI string) to match the Incoming call leg to an inbound dial peer + port ~ Attempts to match the configured dial-peer port to the voice port associated with the incoming call (POTS dial peers only) + dial-peer 0 Will be matched as a last resort Bee A In determining how a router matches inbound dial peers, itis important to note whether the router will match the inbound call leg to a POTS or VoIP dial peer. Matching occurs in the following manner: Inbound POTS dial peers are associated with the incoming POTS call legs of the originating router or gateway. Inbound VoIP dial peers are associated with the incoming VoIP call legs of the terminating, rouler or gateway The router will match three information elements sent in the call setup message against four configurable dial-peer command attributes. ‘The table lists the three call setup information elements that are known about calls arriving at the gateway. Call Setup Information Elements Call Setup Element Description Called number Dialed Number | This is the call destination dial string, and itis derived from the Identification Service (DNIS) ISDN setup message or the channel associated signaling (CAS) DNis. Calling number Automatic Number | This is a number string that represents the call origin, and its Identification (ANI) derived from the ISDN setup message or the CAS ANI. The ANI is also referred to as the caling line ID (CLID) Voice port This element represents the POTS physical voice port 3.58 IP Telophony Expross(IPTX) v4.0 (© 2006 Cisco Systems, ine.‘When the Cisco IOS router or gateway receives a call setup request, it makes a dial-peer match for the incoming call. This is not digit-by-digit matching; instead, the router uses the full digit string received in the setup request. The router or gateway matches call setup element parameters in the order shown in the table. How the Router or Gateway Matches Inbound Dial Peers step Action 1 “The router or gateway attempts to match the called numberof the call setup request with the configured incoming called number of each dial peer. 2 Ifa match is not found, the router or gateway attempts to match the caling number of the call setup request with the answer address of each dial peer. 3 Ifa match is not found, the router or gateway attempts to match the caling number of the call setup request to the destination patter of each dial peer. 4 “The voice port uses the voice port number associated with the incoming call setup request fo match the inbound call leg to the configured dial-peer port parameter. 5 If multiple cial paers have the same port configured, then the router or gateway matches the first dil peer added to the configuration. 6 I match Is not found in the previous steps, then the default is dial peer 0. Because call setups always include DNIS information, it is recommended that you use the incoming ealled-number command for inbound dial-peer matching. Configuring the incoming called-number command is useful for a company that has a central call center that provides support for a number of products. Purchasers of each product get a unique 1-800 number to call for support. All support calls are routed to the same trunk group that is destined for the call center. When a call comes in, the computer telephony system uses the DNIS to flash the appropriate message on the computer screen of the agent to whom the call is routed. The agent then knows how to customize the greeting when answering the call. It is useful to configure the calling number ANI with the answer-address command when you ‘want to match calls based on the originating calling number. For example, when a company has intemational customers who require foreign language-speaking agents to answer the call, the call can be routed to the appropriate agent based on the country of call origin, ‘You must configure the calling number ANI with the destination-pattern command when you set up the dial peers for two-way calling. In a corporate environment, the head office and the be connected. As long as each site has a VoIP dial peer configured to point to that dial peer. remote sites mu each site, inbound calls from the remote site match agai (© 2006 Cisco Systems, Ine. PSTN Intertace and Voce Dial Peer Configuration 387What Is the Default Dial Peer? This topic describes the default dial peer. PTET SPE ae) = Dial peer 0 may not be deleted, modified, or changed + Fails to negotiate services and applications: — DTMF relay = VAD = TCL applications + Dial peer 0 configuration with inbound VoIP calls: any codec ~ vad enabled ~ no rsvp support ~fax-rate voice + Dial peer 0 configuration with inbound POTS calls: ~ no ivr application When the system does not find # matching inbound dial peer, the router resorts to the default ial peer. The default dial peer is referred to as dial-peer 0. When the default dial peer matches on a VoIP call, the call leg that is set up in the inbound direction uses any supported codec for voice compression, based on the requested codec capability coming from the source router. When a default dial peer matches, the voice path in ‘one direction may have parameters that are different from the voice in the retum direction, This behavior may cause one side of the connection to report good-quality voice while the other side reports poor-quality voice. For example, the outbound dial peer has VAD disabled, but the inbound call leg is matched against the default dial peer, which has VAD enabled. In this example, VAD is on in one direction and off in the return direction, When the default dial peer matches an inbound POTS call leg, there is no default IVR application with the port; as a result, the user gets a dial tone and proceeds with dialed digits. The use of @ catch-all dial peer that matches all calls can prevent the use of the default dial peer and send any matches to a default location, such as the operator or an automated attendant Note Default dial peers are used for inbound matches only. They are not used to match outbound calls that do not have a dial peer configured Dial peer 0 (pid:0) has a default configuration that cannot be changed. The default dial peer 0 fails to negotiate nondefault capabilities, services, and applications such as these: = Nondefault voice-network capabilities: DTMF relay, no VAD, and so on © Direct inward dialing (DID) = Toolkit Command Language (TCL) applications 358 P Telephony Express (PTX) v4.0, (© 2006 Cisco Systems, nc.Dial peer 0 for inbound VoIP peers has the following configuration: = any codec = ip precedence 0 m= vad enabled = no rsvp support m fax-rate service Dial peer 0 for inbound POTS peers has the following configuration: = no ivr application Pete ea Prone jz | |, Matches Dia aiches Oil Pee ia 0t0" | |_Peer oinbouns “Outbound | + First, check for incoming called-number command that matches the 2010. + Second, check for answor-address command that matches the 1234. + Third, check for destination-pattorn command that matches the 1234. + Fourth, check for a dial peer with port 1/011 + Fifth, dial peer 0 will be used because no other option exists. # Example In this figure, CMERouter! has no defined matching dial peer. The default dial peer 0 is matched inbound and dial peer 30 is matched outbound. (© 2006 Cisco Systems, nc. PSTN interlace and Voice Dial Peer Configuration 3-59Matching Outbound Dial Peers This topic describes how the router matches outbound dial peers. Matching Outbound Dial Peers Destination pattern is matched based on longest number match, Example 1: Dialed number 555-1234 wil match dial peer 4 Example 2: Dialed number 555-1235 wil match dial peer 3. Example 3: Dialed number 555-2000 will match dial peer 2. Example 4: Dialed number 851-1234 will match dial peer 1 ee Outbound dial peer matching is completed on a digit-by-digit basis. Therefore, the router or gateway checks for dial peer matches after receiving each digit, then routes the call when a full match is made, ‘The router or gateway matches outbound dial peers in the order shown in the table. How the Router or Gateway Matches Outbound Dial Peers step Action 1 The router or gateway uses the destination-pattem command under the dial peer to determine how to route the cal ‘The destination-pattern command routes the cal inthe following manner ‘= On POTS dial peers, the port command forwards the call = On VoIP dial peers, the session target command forwards the call 3 Use the show dialpian number string command to determine which dial peer is matched to a specific dialed string, This command displays all matching dial peers in the order in which they are used Example In this figure, dial peer 1 matches any digit string that has not matched other dial peers more specifically. Dial peer 2 matches any seven-digit number in the 2000 and 3000 range of numbers starting with 555. Dial peer 3 matches any seven-

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