Cisco SIP Trunk Operations (DTSIP)

Name: Cisco SIP Trunk Operations (DTSIP)
Price: 4295.00 USD
Availability: InStock
Rating: 4.5 (1200 reviews)

Price
$4,295.00 USD

Duration
5 Days

Cisco Learning Credit (CLC) Eligible

Delivery Methods
Virtual Instructor Led
Private Group

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Course Overview

SIP Trunk Operations (DTSIP) is a 5-day instructor-led course that is intended for Cisco collaboration administrators who need to understand the features and functionality of the SIP protocol, as implemented in Cisco’s Collaboration deployments.

This course has been updated to include the latest Cisco Unified Communications Manager version 15 servers, adding cutting-edge content and labs designed to enhance practical skills and knowledge.

New Content and Labs:

CUBE High Availability: Introduction and configuration of high availability features in CUBE to ensure continuous service in enterprise environments.
Calling Privileges Configuration: Detailed lab on configuring and managing calling privileges to ensure proper access and restrictions within the network.
Introduction to AI Troubleshooting using ChatGPT: Utilize AI tools like ChatGPT to analyze and interpret SIP debug messages and call trace, enhancing troubleshooting effectiveness.
DSP Functionality, Codecs, and Codec Complexity: Explore the digital signal processing aspects, different codecs used, and their complexities.
Advanced SIP Traces Analysis: More in-depth labs on examining and understanding a variety of SIP traces, helping participants master the nuances of SIP communications.
Introduction to Basic DevOps for Cisco Environments:
Infrastructure as Code with Ansible: Learn how to create and manage Cisco routers using Ansible, introducing the basics of automation and orchestration in network configuration, Github, and Cisco Modeling Labs to simulate and test network configurations and scenarios in a virtual environment.

Course Objectives

Examine and understand the purpose of SIP requests, responses, and SDP
Configure SIP trunks and SIP Profiles on Cisco Unified Communication Manager (CUCM)
Configure SIP call routing on Cisco SIP Proxy (CUSP)
Configure URI Call routing on both CUCM and Session Border Controllers (CUBEs)
Configure SIP CUBEs using a variety of features, including translation-profiles, patterns-maps, server groups, provision policies
Gather SIP traces from servers, CUBEs, routers, phones, endpoints
Diagnose and resolve SIP call routing issues, including one-way audio, misconfiguration, and many other commonly encountered ’real world’ issues
Configure and troubleshoot SIP throughout their collaboration enterprise

Who Should Attend?

Cisco Unified Communications Manager
Professionals with CCNA Voice and/or CCNP Voice Certification
Customers that need to better understand the SIP protocol

Top-rated instructors: Our crew of subject matter experts have an average instructor rating of 4.8 out of 5 across thousands of reviews.
Authorized content: We maintain more than 35 Authorized Training Partnerships with the top players in tech, ensuring your course materials contain the most relevant and up-to date information.
Interactive classroom participation: Our virtual training includes live lectures, demonstrations and virtual labs that allow you to participate in discussions with your instructor and fellow classmates to get real-time feedback.
Post Class Resources: Review your class content, catch up on any material you may have missed or perfect your new skills with access to resources after your course is complete.
Private Group Training: Let our world-class instructors deliver exclusive training courses just for your employees. Our private group training is designed to promote your team’s shared growth and skill development.
Tailored Training Solutions: Our subject matter experts can customize the class to specifically address the unique goals of your team.

Course Prerequisites

CCNA Voice or equivalent knowledge or,
Knowledge gained from attending prerequisite courses: VFCC or ACUCM w/AUC

Agenda

Module 1 Examining Collaboration Solutions

Section 1: Exploring the Path to Collaboration – CLFNDU
Describe On-Premise deployment
Examine cloud deployments
Examine collaboration endpoints

Module 2: Examining SIP Call Signaling and Codecs

Section 4: Exploring Codecs and Call Signaling- CLFNDU Section 2: Exploring Call Signaling over IP Networks -CLCOR
Describe SIP call signaling, voice and video codecs, RTP and RTCP
Describe the Call Setup and Teardown Process
Describe SIP Call Signaling for Call Setup and Teardown
Explore Media Streams at the Application Layer
Compare Audio Codecs
Compare Video Codecs

Module 3: Analyzing and Troubleshooting SIP Signaling

Section 1: Analyzing and Troubleshooting Signaling and Media Protocols- CLACCM
Analyze and troubleshoot SIP and media protocols
Examine the characteristics and features of SIP
SIP Trunking Considerations
SIP Troubleshooting Tools
Configuring SIP Traces using RTMT
Using Wireshark and TranslatorX to read SIP debugs and traces
Using Cisco Support Tools like CUBE DNA and Collaboration Analyzer to troubleshoot SIP calls

Module 4: Configuring Cisco SIP Trunks and Proxy

Examine and configure SIP Proxy to route calls and CUCM SIP trunk features and capabilities
Configuring SIP trunks to provide call routing
Examining CUCM SIP trunk settings and understanding their purpose
Examining CUCM SIP Profile settings and understanding their purpose
Examining SIP Proxy Call Processing
Configuring SIP Proxy to manage enterprise calls

Module 5 Implementing SIP URI Calling on CUCM

Section 11: Implementing URI Calling in Cisco Unified Communications Manager- CLACCM
Implementing URI calling in CUCM for calls within a cluster and between clusters
Provide an overview of URI call routing in CUCM
Describe Directory URIs in CUCM
Describe the URI call routing process in CUCM
Describe how CUCM routes SIP URI calls to other call control systems using SIP route patterns and SIP trunks
Describe what needs to be considered when implementing URI call routing in CUCM

Module 6: Deploying ILS and GDPR

Section 13: Examining Global Dial Plan Replication- CLACCM
Describe how to implement ILS between CUCM clusters and enable GDPR This lesson
Describe global dial plan issues
Describe the characteristics of ILS and its services
Describe the components of GDPR and their interaction
Describe how calls are routed using GDPR
Describe how to implement PSTN backup for intercluster calls when using GDPR

Module 7: Deploying Cisco SIP Voice Gateways

Section 9: Describing the Cisco ISR as a Voice Gateway – CLFNDU
Describe the function, purpose, and configuration of the Cisco SIP ISR gateway
Describe Cisco Voice Gateways
Describe SIP gateways
Describe Call Legs and Dial Peers
Describe Digital Signaling Processors
Explore the DSP Calculator

Module 8: Configuring Session Border Controllers (CUBEs)

Section 14: Configuring and Troubleshooting Cisco Unified Border Element- CLACCM
Configure and troubleshoot Cisco Unified Border Element (CUBE)
Describe the Cisco Unified Border Element
Describe the call-routing logic in CUBE for numeric and URI calls
Understand the advanced options for CUBE
Describe how to manipulate SIP header and SDP elements in CUBE using SIP profiles
Understand CUBE signaling and media bindings

Module 9: Configuring Additional SIP CUBE Settings

Section 8: Implementing Voice Gateways – CLCOR
Describe how to implement digit manipulation, Early Offer, and OPTIONS on a Cisco SIP CUBE
Configuring Voice translation profiles on CUBE
Configuring SIP Early offer on the CUBE
Configuring MTP on SIP Trunk to support early offer
Configuring SIP OPTIONS keepalives on CUBE

Module 10: Configuring CUBE based URI Call Routing

Configuring inbound URL dial-peer matching
Configuring outbound URL dial-peer matching
Configuring SIP Calling and Connected Party Info
Configuring Provisioning Policies
Normalizing SIP Messages

Module 11: Configuring the Summary Lab

Configuring SIP trunks, CUBE, dial plan, and a variety of other settings students learned during the class
There is a list of requirements that students will fulfill and SIP related problems that students will solve
This lab helps students solidify concepts and demonstrates their proficiency in building SIP deployments