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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.
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
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
Course Outline
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