DTSIP - SIP Trunk Operations
The course begins with an examination of SIP Request and Response messages, their purpose, their meanings. We examine the Session Description Protocol (SDP) offers and answers. We explain SIP early offer and SIP early media. We also cover the purpose and configuration of Media Termination Points (MTP) and transcoders in our SIP deployments. We examine the headers that makeup all SIP messages.
Next, we examine the features and capabilities of CUCM SIP trunks. We cover the purpose of options available on the CUCM SIP Profiles that are used for trunks and line-side endpoint configurations. We will configure SIP URI dialing on CUCM. We will use ILS, GDPR, and an SME server to dynamically distribute the dial plan among multiple CUCM clusters. We configure the Cisco SIP Proxy to route enterprise calls.
We will configure Session Border Controllers (CUBES) for a variety of connective purposes. We will demonstrate how the use of E.164 Pattern Maps and Server-Groups will significantly improve and simplify the CUBE configuration. We examine the call routing logic of both inbound and outbound dial peers. We configure Voice Translation Profiles and Dial Peer Groups. We configure URI Call Routing on the CUBE and demonstrate how Provisioning Policies allow administrators to select outbound dial peers based on inbound dial-peer matching. We show you how to configure SIP Normalization on both the CUBE and CUCM, as well as how to configure the SIP OPTIONS ping keepalive feature.
In this course, we will spend an extensive amount of time Troubleshooting SIP calls. We will demonstrate many ways to collect SIP Traces and Debugs and show you how to use diagnostic programs that are available to examine and understand the various SIP debug and trace output.
Finally, you will configure a summary lab that will challenge you to use the knowledge and skills you will have learned throughout the course. You will configure an end-to-end SIP solution using multiple CUCM clusters and CUBEs. You will fulfill a list of SIP configuration requirements like what you will encounter in your real-world collaboration deployment.
After this course, students will be able to:
- 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