ICCCCHCurrent Technologies CLCCH-ICCC1.0<p>Upon completing this course, the student will be able to meet these objectives: </p> <ul> <li>Configuring Gateway Voice Ports</li><li>SIP Protocols</li><li>Configuring VoIP Call Legs</li><li>Implementing a Dial Plan</li><li>Configuring Cisco Unified Communication Manager 11.x</li><li>Deploying Cisco VCUBE</li><li>Implementing Cisco Unified Border Element</li><li>High Availability on ISR G2, ISR 4k, and ASR</li><li>Security on Cisco Unified Border Element</li><li>Monitoring and Troubleshooting on Cisco Unified Border Element</li></ul><p>Understand network fundamentals</p><ul> <li>Network Video Engineer</li><li>Voice / UC / Collaboration / Communications Engineer</li><li>Collaboration Tools Engineer</li><li>Collaboration Sales / Systems Engineer</li></ul><h4>Module 0: Introduction</h4><ul> <li>Module Introduction<ul> <li>Topic List</li></ul></li><li>Lesson 1: Introductions<ul> <li>Topic List</li><li>Learner Skills and Knowledge</li><li>Couse Goals</li><li>WebEx Basics</li><li>General Administration</li><li>Introductions</li></ul></li><li>Lesson 2: SIP Trunking<ul> <li>SIP Trunking Overcomes TDM Barriers</li><li>Why does an enterprise need an SBC?</li><li>Primary CUBE Differentiators (1)</li></ul></li><li>Module Summary</li></ul><h4>Module 1: Introduction to Voice Gateways</h4><ul> <li>Module Introduction<ul> <li>Topic List</li></ul></li><li>Lesson 1: Gateways<ul> <li>PSTN Access Methods</li><li>TDM Gateway vs. Cisco UBE</li></ul></li><li>Gateway Functionality (1)</li><li>VoIP Signaling Protocols (1)</li><li>Gateway Deployment Example</li><li>Cisco Unified Communications Deployment Models<ul> <li>Single-Site Deployment (1)</li><li>Multisite WAN with Centralized Call Processing (1)</li><li>Multisite WAN with Distributed Call Processing (1)</li><li>Gateway Hardware Platforms<ul> <li>Cisco 4000 Series ISR Portfolio</li><li>Gateway Hardware - Voice Gateways</li><li>Voice Gateway Overview</li><li>Voice Gateway Call Legs</li></ul></li><li>Cisco Unified Border Element</li><li>Summary</li></ul></li><li>Lesson 2: Gateway Call Routing Components<ul> <li>Inbound and Outbound Dial Peers<ul> <li>Most Prevalent Dial-Peer Types</li><li>POTS and VoIP Dial Peers</li><li>VoIP Dial Peers</li><li>VoIP Dial Peer Examples</li></ul></li><li>IP and Call Routing Comparison</li><li>Call Routing</li><li>Call Legs (1)</li></ul></li><li>Lesson 3: Dial Peer Matching<ul> <li>Use of String Matching</li><li>String-Matching Characters</li><li>Number-Matching Examples</li><li>Matching Inbound Dial Peers (1)</li><li>Matching Outbound Dial Peers (1)</li><li>Summary</li></ul></li><li>Lesson 4: Introduction to Voice Gateways<ul> <li>Verifying Voice Ports (1)</li><li>Voice Codecs (1)</li><li>Voice Codec Packet Rates and Payload Sizes</li><li>Voice Quality Evaluation</li><li>Codec Quality</li><li>Evaluating Overhead (1)</li><li>Per-Call Bandwidth Using Common Codecs</li></ul></li><li>Lesson 5: Digital Signal Processors<ul> <li>Digital Signal Processors (1)</li><li>DSP Modules</li><li>DSP Module Comparison</li></ul></li><li>Lesson 6: Codec Complexity<ul> <li>Codec Complexity (1)</li><li>Packet Voice DSP Module Conferencing</li></ul></li><li>Lesson 7: Configuring DSPs<ul> <li>Configuring DSPs for Voice Termination (1)</li><li>Codec Complexity Configuration</li><li>Configuring DSP Resources for Transcoding, Conferencing, and MTP (1)</li><li>Transcoding and Conferencing Example</li><li>Verifying DSPs (1)</li><li>Summary</li></ul></li><li>Module Summary</li></ul><h4>Module 2: Gateway Dial Plans</h4><ul> <li>Module Introduction<ul> <li>Topic List</li></ul></li><li>Lesson 1: VoIP Overview<ul> <li>VoIP and Traditional Telephony Comparison<ul> <li>VoIP Components</li><li>VoIP Media Transmission Overview</li></ul></li><li>Real-Time Transport Protocol<ul> <li>Real-Time Transport Control Protocol</li><li>Secure RTP</li></ul></li><li>VoIP Media Considerations</li><li>Voice Activity Detection Overview</li></ul></li><li>Lesson 2: SIP Signaling Protocol<ul> <li>SIP Architecture Overview<ul> <li>SIP Signaling</li><li>SIP Architecture Components</li><li>SIP Servers</li><li>SIP Architecture Examples</li><li>SIP Direct Call Flows</li></ul></li></ul></li><li>Lesson 3: SIP Addressing<ul> <li>SIP Address Types</li><li>Address Registration</li><li>Address Resolution</li></ul></li><li>Lesson 4: Codecs in SIP<ul> <li>Session Description Protocol<ul> <li>SDP Examples</li><li>Delayed Offer</li><li>Early Offer</li><li>Early Media (1)</li></ul></li></ul></li><li>Lesson 5: Configuring Basic SIP<ul> <li>Basic SIP Configuration Overview<ul> <li>User Agent Configuration</li><li>Dial Peer Configuration</li><li>Basic SIP Configuration Example</li></ul></li></ul></li><li>Lesson 6: Configuring SIP ISDN Support<ul> <li>SIP SRTP Support Overview<ul> <li>SIPS Global and Dial Peer Commands</li><li>SRTP Global and Dial Peer Commands</li><li>SIPS and SRTP Configuration Example</li></ul></li></ul></li><li>Lesson 7: Customizing SIP Gateways<ul> <li>SIP Gateways Tuning Overview<ul> <li>SIP Transport</li><li>SIP Source IP Address and UA Timers</li><li>SIP UA Timers</li><li>SIP Early Media</li></ul></li><li>Gateway-to-Gateway Configuration Example</li><li>UA Example</li></ul></li><li>Lesson 8: Verifying SIP Gateways<ul> <li>show sip-ua Command Overview<ul> <li>SIP-UA General Verification</li><li>SIP-UA Registration Status and Timers</li><li>SIP-UA Call Information</li></ul></li><li>SIP Debugging Overview</li><li>Examining the INVITE Message</li><li>Examining the 200 OK Message</li><li>Examining the BYE Message</li></ul></li><li>Lesson 9: Audio Clarity Requirements<ul> <li>Audio Clarity Factors</li><li>Delay Sources</li><li>Delay Types</li><li>Acceptable Delay (G.114)</li><li>Jitter</li><li>Packet Loss</li><li>Summary of QoS Objectives</li></ul></li><li>Module Summary</li></ul><h4>Module 3: Cisco Unified Communication Manager (CUCM)</h4><ul> <li>Module Introduction<ul> <li>Topic List</li></ul></li><li>Lesson 1: Configuring CUCM for Gateways and Trunks<ul> <li>Check CUCM Version</li><li>Cisco UCM Audio Codec Preference List (1)<ul> <li>Audio Codec Preference List - Copy Existing</li><li>Audio Codec Preference List - New List</li><li>Audio Codec Preference List - Name and Order</li></ul></li></ul></li><li>Cisco UCM Region Configuration</li><li>Region Configuration - Region Relationships</li><li>Device Pool Configuration - Add New Device Pool (1)</li><li>Annunciator Configuration</li><li>Conference Bridge Configuration</li><li>Media Termination Point Configuration</li><li>Music on Hold Server Configuration<ul> <li>Music on Hold Service (IP Voice Media Streaming App) Parameter Settings (1)</li><li>Music on Hold Service (Duplex Streaming) Parameter Settings</li><li>Media Resource Group Configuration (1)<ul> <li>Media Resource Group List Configuration</li></ul></li></ul></li><li>Lesson 2: SIP Trunk Configuration<ul> <li>SIP Trunk Security Profile Configuration (1)</li></ul></li><li>SIP Profile Configuration (1)</li><li>SIP Trunk to Cisco CUBE Configuration (1)</li><li>Reset SIP Trunks</li><li>Path Selection Configuration Elements in Cisco Unified Communications Manager</li><li>Create Route Pattern to get to the CUBE</li><li>Module Summary</li></ul> <h4>Module 4: Configuring Cisco Unified Border Element (CUBE)</h4><ul> <li>Module Introduction<ul> <li>Topic List</li></ul></li><li>Lesson 1: What Does a Session Border Controller (SBC) Do?<ul> <li>What Is Driving SBC Adoption?</li><li>CUBE&rsquo;s Primary Strategic Differentiators</li><li>Cisco Unified Border Element &ndash; Router Integration</li><li>CUBE Offers Architectural Choice and Flexibility</li><li>CUBE Is Scalable</li><li>CUBE Is Robust and Reliable</li><li>CUBE Session Capacity Summary</li><li>CUBE Interoperability</li><li>CUBE Is Easy to Monitor and Manage</li><li>SIP Trunking to Cisco WebEx</li><li>Using CUBE in Your Contact Center Environment</li><li>CUBE Licensing (1)</li><li>Cisco Session Management &amp; CUBE</li><li>CUBE/vCUBE Deployment Scenarios</li><li>The Centralized Model</li><li>The Distributed Model</li><li>The Hybrid Model</li></ul></li><li>Lesson 2: CUBE Call Flow<ul> <li>CUBE Call Processing</li><li>Cisco Unified Border Element Basic Call Flow</li><li>Transitioning to Centralized SIP Trunking...</li><li>Steps to transitioning...<ul> <li>Step 1: Configure CUCM to route calls to the edge SBC</li><li>Step 2: Get details from SIP Trunk provider</li><li>Step 3: Enable CUBE Application on Cisco routers</li><li>Step 4: Configure Call routing on CUBE</li></ul></li></ul></li><li>WAN Dial-Peer Configuration</li><li>LAN Dial-Peer Configuration</li><li>Lesson 3: CUBE Dial-Peers Call Routing<ul> <li>Understanding Dial-Peer Matching Techniques</li><li>Understanding Inbound Dial-Peer Matching Techniques (1)</li><li>Understanding Outbound Dial-Peer Matching Techniques (1)</li></ul></li><li>Lesson 4: CUBE Advanced Call Routing<ul> <li>Additional Headers for Outbound Dial-Peer Matching</li><li>Introducing Outbound Dial-peer Provision Policy</li><li>Dial-peer Provision Policy Configuration (1)</li><li>Dial-peer Provision Policy Example &ndash; Match on FROM (1)</li><li>Destination Dial-peer Group</li><li>Destination Dial-peer Group Configuration</li><li>Destination Server Group</li><li>Multiple Destination-Patterns Under Same</li><li>Multiple Incoming Patterns Under Same</li></ul></li><li>Lesson 5: Media Manipulation<ul> <li>Audio Transcoding and Transrating</li><li>Configuration for SCCP based Transcoding</li><li>Configuration for LTI based Transcoding</li></ul></li><li>Lesson 6: External/PSTN Call Recording<ul> <li>External/PSTN Call Recording Options</li><li>CUBE Controlled Recording Option &ndash; Media Forking</li><li>Audio only Media Forking for an Audio/Video Call</li><li>CUBE Controlled Recording Option &ndash; SIPREC</li></ul></li><li>Lesson 7: Call Admission Control<ul> <li>Call Admission Control Based on Total Calls, CPU and Memory usage (1)</li><li>Call Admission Control at the edge</li><li>Call Admission Control based on Call spikes (1)</li><li>Call Admission Control based on Bandwidth</li></ul></li><li>Lesson 8: Multiple Non-Authenticated SIP Trunks on a CUBE<ul> <li>Non-Authenticated SIP Trunking to more than one Service Provider</li></ul></li><li>Lesson 9: Multiple Authenticated/Registered SIP Trunks on a CUBE<ul> <li>Multiple Instances of SIP-UA on a CUBE</li><li>Introducing Tenants on CUBE</li><li>&ldquo;Voice class Tenant&rdquo; Overview</li><li>Authenticating Multiple trunks with same Realm</li><li>Configuring Voice Class Tenant</li></ul></li><li>Module Summary</li></ul><h4>Module 5: Configuring High Availability</h4><ul> <li>Module Introduction<ul> <li>Topic List</li></ul></li><li>Lesson 1: High Availability<ul> <li>CUBE High Availability Options<ul> <li>CUBE HA Design Considerations on ISR-G2 for Box-to-Box Redundancy (1)</li></ul></li></ul></li><li>CUBE Configuration on ISR-G2 Box-to-Box Redundancy (1)</li><li>CUBE HA Design Considerations on ASR1K/ISR-4K/vCUBE for Box-to-Box Redundancy (1)</li><li>CUBE Configuration on ASR1K-ISR-4K/vCUBE Box-to-Box Redundancy (1)</li><li>Additional Supported Options for CUBE HA</li><li>ASR B2B Redundancy: PROTECTED MODE</li><li>CUBE SIP Trunk Monitoring with OOD Options Message (1)</li><li>OOD Options Ping Keepalive Enhancement (1)</li><li>SIP Trunk to TDM PSTN Failover</li><li>Lesson 2: MMoH<ul> <li>Multicast MoH to Unicast MoH Conversion- CUBE</li></ul></li><li>Module Summary</li></ul><h4>Module 6: Security</h4><ul> <li>Module Introduction<ul> <li>Topic List</li></ul></li><li>Lesson 1: CUBE Security<ul> <li>Five Layers of Security in CUBE</li><li>Cube Security Best Practices Summary</li><li>Topology/Addressing Hiding</li><li>SIP Trunk to ITSP</li><li>IP Trust List for Signaling</li><li>Toll Fraud Mitigation</li><li>Configure Call Routing on CUBE</li></ul></li><li>Understanding Dial-Peer Matching Techniques: LAN &amp; WAN Dial-Peers</li><li>WAN Dial-Peer Configuration</li><li>LAN Dial-Peer Configuration</li><li>ACLS Applied on WAN Interfaces<ul> <li>SIP Listening Port Protection</li><li>Close Unused Session Transport Mechanisms</li></ul></li><li>NBAR to Protect Against SIP Flooding and UDP Attacks at Opened RTP Ports</li><li>Firewall: General Guidelines<ul> <li>CUBE Firewall Deployment Scenarios</li></ul></li><li>Lesson 2: SIP TLS Support with SRTP<ul> <li>Secure SIP</li><li>SRTP Passthrough Configuration (Unsupported Crypto Suites)</li></ul></li><li>SIP TLS/SRTP Support for Microsoft Skype for Business (Lync) Interop</li><li>Module Summary</li></ul><h4>Module 7: Monitoring and Troubleshooting </h4><ul> <li>Module Introduction<ul> <li>Topic List</li></ul></li><li>Lesson 1: Monitoring and Troubleshooting Cisco Cube<ul> <li>Dialed Number Analyzer (DNA) for CUBE</li></ul></li><li>Input Option - Calling/Called Number (1)</li><li>Input Option- SIP Message (1)</li><li>Lesson 2: SIP Profile Test Tool<ul> <li>SIP Profile Test Tool (1)</li><li>CUBE Monitoring</li><li>Prime Collaboration (1)<ul> <li>Prime Collaboration- Assurance (1)</li><li>Prime Collaboration- Analytics (1)</li></ul></li></ul></li><li>Lesson 3: Troubleshooting<ul> <li>Troubleshooting of Calls<ul> <li>CUBE Debugging</li><li>SIP Debug (1)</li><li>CUBE Per-Call Debugging (PCD) (1)</li></ul></li></ul></li><li>IOS Embedded Packet Capture on ISR-G2 (1)<ul> <li>Pcap</li><li>Export pcap</li><li>Call Setup Failure<ul> <li>IOS Command Output after Outbound Call</li><li>IOS Command Output after Inbound Call Attempt</li></ul></li><li>Debug ccsip message<ul> <li>debug voip ccapi inout</li><li>debug ccsip transport debug ip tcp transaction debug crypto pki</li><li>Troubleshooting (1)</li><li>Verifying Certificates</li></ul></li></ul></li><li>Lesson 4: Serviceability<ul> <li>New CUBE Serviceability Features</li><li>Call History Stats- Graphical or Tabular Form</li></ul></li><li>Total Number of Active Concurrent Calls</li><li>Debugging Made Easier (1)</li><li>Module Summary</li></ul>Upon completing this course, the student will be able to meet these objectives: - Configuring Gateway Voice Ports - SIP Protocols - Configuring VoIP Call Legs - Implementing a Dial Plan - Configuring Cisco Unified Communication Manager 11.x - Deploying Cisco VCUBE - Implementing Cisco Unified Border Element - High Availability on ISR G2, ISR 4k, and ASR - Security on Cisco Unified Border Element - Monitoring and Troubleshooting on Cisco Unified Border ElementUnderstand network fundamentals- Network Video Engineer - Voice / UC / Collaboration / Communications Engineer - Collaboration Tools Engineer - Collaboration Sales / Systems EngineerModule 0: Introduction - Module Introduction - Topic List - Lesson 1: Introductions - Topic List - Learner Skills and Knowledge - Couse Goals - WebEx Basics - General Administration - Introductions - Lesson 2: SIP Trunking - SIP Trunking Overcomes TDM Barriers - Why does an enterprise need an SBC? - Primary CUBE Differentiators (1) - Module Summary Module 1: Introduction to Voice Gateways - Module Introduction - Topic List - Lesson 1: Gateways - PSTN Access Methods - TDM Gateway vs. Cisco UBE - Gateway Functionality (1) - VoIP Signaling Protocols (1) - Gateway Deployment Example - Cisco Unified Communications Deployment Models - Single-Site Deployment (1) - Multisite WAN with Centralized Call Processing (1) - Multisite WAN with Distributed Call Processing (1) - Gateway Hardware Platforms - Cisco 4000 Series ISR Portfolio - Gateway Hardware - Voice Gateways - Voice Gateway Overview - Voice Gateway Call Legs - Cisco Unified Border Element - Summary - Lesson 2: Gateway Call Routing Components - Inbound and Outbound Dial Peers - Most Prevalent Dial-Peer Types - POTS and VoIP Dial Peers - VoIP Dial Peers - VoIP Dial Peer Examples - IP and Call Routing Comparison - Call Routing - Call Legs (1) - Lesson 3: Dial Peer Matching - Use of String Matching - String-Matching Characters - Number-Matching Examples - Matching Inbound Dial Peers (1) - Matching Outbound Dial Peers (1) - Summary - Lesson 4: Introduction to Voice Gateways - Verifying Voice Ports (1) - Voice Codecs (1) - Voice Codec Packet Rates and Payload Sizes - Voice Quality Evaluation - Codec Quality - Evaluating Overhead (1) - Per-Call Bandwidth Using Common Codecs - Lesson 5: Digital Signal Processors - Digital Signal Processors (1) - DSP Modules - DSP Module Comparison - Lesson 6: Codec Complexity - Codec Complexity (1) - Packet Voice DSP Module Conferencing - Lesson 7: Configuring DSPs - Configuring DSPs for Voice Termination (1) - Codec Complexity Configuration - Configuring DSP Resources for Transcoding, Conferencing, and MTP (1) - Transcoding and Conferencing Example - Verifying DSPs (1) - Summary - Module Summary Module 2: Gateway Dial Plans - Module Introduction - Topic List - Lesson 1: VoIP Overview - VoIP and Traditional Telephony Comparison - VoIP Components - VoIP Media Transmission Overview - Real-Time Transport Protocol - Real-Time Transport Control Protocol - Secure RTP - VoIP Media Considerations - Voice Activity Detection Overview - Lesson 2: SIP Signaling Protocol - SIP Architecture Overview - SIP Signaling - SIP Architecture Components - SIP Servers - SIP Architecture Examples - SIP Direct Call Flows - Lesson 3: SIP Addressing - SIP Address Types - Address Registration - Address Resolution - Lesson 4: Codecs in SIP - Session Description Protocol - SDP Examples - Delayed Offer - Early Offer - Early Media (1) - Lesson 5: Configuring Basic SIP - Basic SIP Configuration Overview - User Agent Configuration - Dial Peer Configuration - Basic SIP Configuration Example - Lesson 6: Configuring SIP ISDN Support - SIP SRTP Support Overview - SIPS Global and Dial Peer Commands - SRTP Global and Dial Peer Commands - SIPS and SRTP Configuration Example - Lesson 7: Customizing SIP Gateways - SIP Gateways Tuning Overview - SIP Transport - SIP Source IP Address and UA Timers - SIP UA Timers - SIP Early Media - Gateway-to-Gateway Configuration Example - UA Example - Lesson 8: Verifying SIP Gateways - show sip-ua Command Overview - SIP-UA General Verification - SIP-UA Registration Status and Timers - SIP-UA Call Information - SIP Debugging Overview - Examining the INVITE Message - Examining the 200 OK Message - Examining the BYE Message - Lesson 9: Audio Clarity Requirements - Audio Clarity Factors - Delay Sources - Delay Types - Acceptable Delay (G.114) - Jitter - Packet Loss - Summary of QoS Objectives - Module Summary Module 3: Cisco Unified Communication Manager (CUCM) - Module Introduction - Topic List - Lesson 1: Configuring CUCM for Gateways and Trunks - Check CUCM Version - Cisco UCM Audio Codec Preference List (1) - Audio Codec Preference List - Copy Existing - Audio Codec Preference List - New List - Audio Codec Preference List - Name and Order - Cisco UCM Region Configuration - Region Configuration - Region Relationships - Device Pool Configuration - Add New Device Pool (1) - Annunciator Configuration - Conference Bridge Configuration - Media Termination Point Configuration - Music on Hold Server Configuration - Music on Hold Service (IP Voice Media Streaming App) Parameter Settings (1) - Music on Hold Service (Duplex Streaming) Parameter Settings - Media Resource Group Configuration (1) - Media Resource Group List Configuration - Lesson 2: SIP Trunk Configuration - SIP Trunk Security Profile Configuration (1) - SIP Profile Configuration (1) - SIP Trunk to Cisco CUBE Configuration (1) - Reset SIP Trunks - Path Selection Configuration Elements in Cisco Unified Communications Manager - Create Route Pattern to get to the CUBE - Module Summary Module 4: Configuring Cisco Unified Border Element (CUBE) - Module Introduction - Topic List - Lesson 1: What Does a Session Border Controller (SBC) Do? - What Is Driving SBC Adoption? - CUBE’s Primary Strategic Differentiators - Cisco Unified Border Element – Router Integration - CUBE Offers Architectural Choice and Flexibility - CUBE Is Scalable - CUBE Is Robust and Reliable - CUBE Session Capacity Summary - CUBE Interoperability - CUBE Is Easy to Monitor and Manage - SIP Trunking to Cisco WebEx - Using CUBE in Your Contact Center Environment - CUBE Licensing (1) - Cisco Session Management & CUBE - CUBE/vCUBE Deployment Scenarios - The Centralized Model - The Distributed Model - The Hybrid Model - Lesson 2: CUBE Call Flow - CUBE Call Processing - Cisco Unified Border Element Basic Call Flow - Transitioning to Centralized SIP Trunking... - Steps to transitioning... - Step 1: Configure CUCM to route calls to the edge SBC - Step 2: Get details from SIP Trunk provider - Step 3: Enable CUBE Application on Cisco routers - Step 4: Configure Call routing on CUBE - WAN Dial-Peer Configuration - LAN Dial-Peer Configuration - Lesson 3: CUBE Dial-Peers Call Routing - Understanding Dial-Peer Matching Techniques - Understanding Inbound Dial-Peer Matching Techniques (1) - Understanding Outbound Dial-Peer Matching Techniques (1) - Lesson 4: CUBE Advanced Call Routing - Additional Headers for Outbound Dial-Peer Matching - Introducing Outbound Dial-peer Provision Policy - Dial-peer Provision Policy Configuration (1) - Dial-peer Provision Policy Example – Match on FROM (1) - Destination Dial-peer Group - Destination Dial-peer Group Configuration - Destination Server Group - Multiple Destination-Patterns Under Same - Multiple Incoming Patterns Under Same - Lesson 5: Media Manipulation - Audio Transcoding and Transrating - Configuration for SCCP based Transcoding - Configuration for LTI based Transcoding - Lesson 6: External/PSTN Call Recording - External/PSTN Call Recording Options - CUBE Controlled Recording Option – Media Forking - Audio only Media Forking for an Audio/Video Call - CUBE Controlled Recording Option – SIPREC - Lesson 7: Call Admission Control - Call Admission Control Based on Total Calls, CPU and Memory usage (1) - Call Admission Control at the edge - Call Admission Control based on Call spikes (1) - Call Admission Control based on Bandwidth - Lesson 8: Multiple Non-Authenticated SIP Trunks on a CUBE - Non-Authenticated SIP Trunking to more than one Service Provider - Lesson 9: Multiple Authenticated/Registered SIP Trunks on a CUBE - Multiple Instances of SIP-UA on a CUBE - Introducing Tenants on CUBE - “Voice class Tenant” Overview - Authenticating Multiple trunks with same Realm - Configuring Voice Class Tenant - Module Summary Module 5: Configuring High Availability - Module Introduction - Topic List - Lesson 1: High Availability - CUBE High Availability Options - CUBE HA Design Considerations on ISR-G2 for Box-to-Box Redundancy (1) - CUBE Configuration on ISR-G2 Box-to-Box Redundancy (1) - CUBE HA Design Considerations on ASR1K/ISR-4K/vCUBE for Box-to-Box Redundancy (1) - CUBE Configuration on ASR1K-ISR-4K/vCUBE Box-to-Box Redundancy (1) - Additional Supported Options for CUBE HA - ASR B2B Redundancy: PROTECTED MODE - CUBE SIP Trunk Monitoring with OOD Options Message (1) - OOD Options Ping Keepalive Enhancement (1) - SIP Trunk to TDM PSTN Failover - Lesson 2: MMoH - Multicast MoH to Unicast MoH Conversion- CUBE - Module Summary Module 6: Security - Module Introduction - Topic List - Lesson 1: CUBE Security - Five Layers of Security in CUBE - Cube Security Best Practices Summary - Topology/Addressing Hiding - SIP Trunk to ITSP - IP Trust List for Signaling - Toll Fraud Mitigation - Configure Call Routing on CUBE - Understanding Dial-Peer Matching Techniques: LAN & WAN Dial-Peers - WAN Dial-Peer Configuration - LAN Dial-Peer Configuration - ACLS Applied on WAN Interfaces - SIP Listening Port Protection - Close Unused Session Transport Mechanisms - NBAR to Protect Against SIP Flooding and UDP Attacks at Opened RTP Ports - Firewall: General Guidelines - CUBE Firewall Deployment Scenarios - Lesson 2: SIP TLS Support with SRTP - Secure SIP - SRTP Passthrough Configuration (Unsupported Crypto Suites) - SIP TLS/SRTP Support for Microsoft Skype for Business (Lync) Interop - Module Summary Module 7: Monitoring and Troubleshooting - Module Introduction - Topic List - Lesson 1: Monitoring and Troubleshooting Cisco Cube - Dialed Number Analyzer (DNA) for CUBE - Input Option - Calling/Called Number (1) - Input Option- SIP Message (1) - Lesson 2: SIP Profile Test Tool - SIP Profile Test Tool (1) - CUBE Monitoring - Prime Collaboration (1) - Prime Collaboration- Assurance (1) - Prime Collaboration- Analytics (1) - Lesson 3: Troubleshooting - Troubleshooting of Calls - CUBE Debugging - SIP Debug (1) - CUBE Per-Call Debugging (PCD) (1) - IOS Embedded Packet Capture on ISR-G2 (1) - Pcap - Export pcap - Call Setup Failure - IOS Command Output after Outbound Call - IOS Command Output after Inbound Call Attempt - Debug ccsip message - debug voip ccapi inout - debug ccsip transport debug ip tcp transaction debug crypto pki - Troubleshooting (1) - Verifying Certificates - Lesson 4: Serviceability - New CUBE Serviceability Features - Call History Stats- Graphical or Tabular Form - Total Number of Active Concurrent Calls - Debugging Made Easier (1) - Module Summary3 days3895.005375.00