Description
Who should attend
- Network Video Engineer
- Voice / UC / Collaboration / Communications Engineer
- Collaboration Tools Engineer
- Collaboration Sales / Systems Engineer
Prerequisites
Understand network fundamentals
Course Objectives
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 Element
Outline: Installing and Configuring Cisco SUBE/Gateways (ICCC)
Module 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)
- Inbound and Outbound Dial Peers
- 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
- VoIP and Traditional Telephony Comparison
- Lesson 2: SIP Signaling Protocol
- SIP Architecture Overview
- SIP Signaling
- SIP Architecture Components
- SIP Servers
- SIP Architecture Examples
- SIP Direct Call Flows
- SIP Architecture Overview
- 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)
- Session Description Protocol
- Lesson 5: Configuring Basic SIP
- Basic SIP Configuration Overview
- User Agent Configuration
- Dial Peer Configuration
- Basic SIP Configuration Example
- Basic SIP Configuration Overview
- 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
- SIP SRTP Support Overview
- 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
- SIP Gateways Tuning Overview
- 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
- show sip-ua Command Overview
- 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 High Availability Options
- 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)
- Troubleshooting of Calls
- 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 Summary