SIP Trunk Optimization (SIPTO)

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About this Course

SIP Trunk Optimization (SIPTO) is a 5-day instructor led course that is intended for experienced administrators who need to better understand the functions and features of SIP, as used within Cisco Unified Communications Manager.  The course will begin with an in depth examination of how SIP is used to establish, manage, and terminate sessions over an IP network. Next, it examines the various types of SIP trunks available within Unified Communications Manager. SIP Trunk Optimization will also cover the configuration that connects users to all devices (phone, mobile, tablet, video) via one step click to call addressing. The course applies troubleshooting techniques within the daily hands-on labs, preparing each student to incorporate these tools inside their own organization.

This course explains exactly what SIP is, what options are available, and how it is used to facilitate communication between various endpoints, such as, Cisco Session Border Element (CUBE) Cisco SIP Proxy server and the PSTN. Students will learn how to use troubleshooting tools to identify and solve a variety of problems that arise during deployment as well as during day to day operations. The course will also use URI ’click to call’ device independent user addressing the SIP Intercluster Lookup Service (ILS) and Global Dial Plan Replication (GDPR) features to dynamically distribute the dial plan throughout the enterprise. 

Sunset Learning developed this course to provide additional training for those who already have a basic understanding of Cisco Unified Communications Manager and are looking to optimize their SIP trunk infrastructure. This is not a certification or implementation course.  

Audience Profile

This course is intended for students who have general knowledge about:

  • Cisco Unified Communications Manager
  • Anyone who has their CCNA Voice and/or is working towards their CCNP Voice Certification
  • Anyone that needs to better understand SIP trunks

At Course Completion

After completing this course, students will be able to...

  • Define the technology components of SIP trunks
  • Create and configure SIP into the Cisco Unified Communications Managers’ Architecture
  • Identify SIP Trunk requirements for PSTN Access service
  • Configure SIP trunks to communicate with Cisco gateways
  • Diagnose and resolve SIP to gateway connectivity and deployment issues
  • Implement Cisco Unified Border Element for SIP-based IP PSTN connectivity
  • Diagnose and resolve CUBE to SIP connectivity and deployment issues
  • Implement Cisco Unified SIP Proxy (CUSP) for Intercluster communication
  • Diagnose and resolve CUSP connectivity and deployment issues
  • Create URI click to call dynamic dial plans
  • Understand how to configure the Intercluster Lookup Service (ILS) and Global Dial Plan Replication (GDPR) features to distribute dynamic dial plan information in an enterprise network.
  • Master troubleshooting techniques: IOS debugs, Real Time Monitoring Tool, Traces, and TranslatorX


To fully benefit from this course, students should have the following prerequisite skills and knowledge: 

  • CCNA Voice or equivalent knowledge, or
  • Knowledge gained from attending prerequisite courses: ICOMM, ACUCM, and SLI’s VFCC



Course Outline

Module 1- What is SIP

  • User Agents
  • SIP Request Methods from RFC 3261 
  • Additional SIP Request Methods 
  • SIP Headers

Module 2 - SIP Call Setup

  • Basic SIP Call Setup with Unified CM and CUBE 
  • Media Negotiation 
  • Session Description Protocol 
  • Media Negotiation - Early Offer and Delayed Offer 
  • Early Media

Module 3 - SIP Trunks PSTN

  • SIP Trunks for PSTN - Access Perception vs. Reality
  • SIP Trunk for PSTN - What to ask for, what to expect
  • Deploying a Cisco Unified Border Element (CUBE)
  • Centralized and Distributed SIP Trunk Models
  • SIP Trunking Migration Plan

Module 4 - Unified CM SIP Trunk

  • The Unified CM SIP Trunk
  • Key Features
  • CUBE Basics and Usage in SIP Calls
  • SIP Trunk Features and Operation
  • Redundancy and Load Balancing
  • Media Resources
  • Called and Calling Party Number Normalization and SIP Trunks

Module 5 - Unified CM SIP Trunk Service Types

  • Session Management Edition
  • Inter-company Media Engine
  • Service Advertisement Framework/Call Control
  • Discovery
  • Extension Mobility Cross Cluster
  • E2E RSVP Trunks

Module 6 - Unified CM - SIP Intercluster Lookup Service

  • User Addressing via URI
  • Inter-cluster Lookup Service
  • Hierarchical Routing & Route String Summarization
  • ILS enhancements in Unified CM 10 - adding Numbers
  • Hierarchical Routing for Numbers and URIs in ILS
  • Route Pattern support in ILS
  • Class Of Service

Module 7 - Unified Cisco SIP Proxy

  • Configuring Cisco Unified SIP Proxy 
  • Editing the General Settings for a Network
  • Viewing, Adding, Moving, and Deleting Rules for a Trigger
  • Observe and Configure SIP Server Groups
  • Observe and Configure Route Groups
  • Observe and Configure Route Tables
  • Configuring Call Admission Control

Module 8 - SIP Troubleshooting Tools 

  • Real Time Monitoring Tool - Session Trace
  • TranslatorX Tool 
  • Wireshark
  • Unified CM Trace Configuration 
  • Trace Collection
  • CUBE Trace Configuration
  • IOS SIP Debugs