CSCGW – Cisco SIP, CUBEs and Gateways

In this course, you will focus on the legacy gateway and router portions of IP Telephony. You will gain extensive experience with the configuration of legacy analog telephony technologies such as Foreign Exchange Station (FXS), Foreign Exchange Office (FXO), and Primary Rate Interface (PRI). In addition to legacy technologies you will gain hands on experience with CUBE and SIP protocols. You will build a working Cisco Unified Communications Manager which will support all major gateway protocols such as MGCP, H.323, and SIP. Troubleshooting will be addressed as a gateway level including common debug techniques and commands.

You’ll gain an understanding of converged voice and data networks as it relates to gateway design and deployment. You will gain comprehensive hands-on experience configuring and deploying Gateways, CUBEs, Quality of Service, and troubleshooting in VoIP networks.

In addition to the knowledge and skills required to integrate gateways into an enterprise VoIP network, youÍll learn how to build and test sophisticated IP telephony dial plans that use both CUCM Dial Plan and Dial Peers at an IOS level which can be used as a template for a real deployment.

The course includes a comprehensive study of Quality of Service (QoS), in which youÍll learn to configure QoS to support real-time traffic.

A Global Knowledge Exclusive:æ You Getƒ

  • Enhanced content that exceeds standard authorized Cisco content
  • Only course dedicated to specific Gateway technologies and Quality of Service
  • World-Class Certified Cisco Systems Instructors

Why Take CSCGW from Global Knowledge?
Every pod has internal and external phones, and just like in a real network, the same simulated public switched telephone network (PSTN) is accessible through all clusters providing failover scenarios for bandwidth and connectivity problems. To more accurately reflect real-world scenarios, you will configure the gateway connections to simulated PBX systems.

We have set ourselves apart from other Cisco training providers by enhancing our CSCGW hands-on labs to include a real dial plan and Class of Service for calling out to the PSTN. Our voice network labs use the latest hardware and software so you will gain experience with the recent stable IOS release (15.X IOS M currently). Plus, each pod contains the following gateway cards for student configuration: 2xFXS, 2xFXO, and 2xT1 ports (PRI and T1-CAS) as well as serial ports for WAN connectivity. All of our IP telephony courses provide a realistic simulated PSTN accessible through both PRI and FXO ports. You will build and test a real dial plan including:

´æthree-digit service codes: 411, 511, and so forth
´æseven-digit local numbers
´æ10-digit local numbers
´æ11-digit long distance numbers
´æInternational numbers
´æConfigure and test all dial peers as appropriate


  • VoIP, components of a VoIP network, VoIP protocols, special requirements for VoIP calls, and Codecs
  • Configure gateway interconnections to support VoIP and PSTN callsæ
  • Basic signaling protocols used on voice gateways
  • Configure a gateway to support calls using different call control and signaling protocolsæ
  • Define a dial plan, describing the purpose of each dial plan component, and implement a dial plan on a voice gatewayæ
  • Implement a Cisco Unified Border Element (CUBE) gateway to connect to an Internet Telephony Service Provider
  • Investigate the use of various traditional telephony connections, such as FXS, FXO, E&M, T1 (CAS and PRI), and E1 (CAS and PRI)æ
  • Configure and troubleshoot Cisco’s new ISR routers and explore their DSP configuration (PVDM3 cards)æ
  • Configure H.323 gateways and review their functions and operation
  • Configure Session Initiation Protocol (SIP) and Media Gateway Control Protocol (MGCP)
  • Experience G.711, and G.729 voice coding schemesæ
  • Configure Call Admission Control three different waysæ
  • Configure proper Caller IDæ
  • Experience real-world connections to PBXs, and the PSTN
  • Configure your router/gateway equipment to connect to our public dial plan network using different call control protocols and procedures


Target Audience

Network engineers, architects, and support staff who:

  • Maintain and configure voice and data network devicesæ
  • Are considering various methodologies to implement VoIPæ
  • Require a fundamental understanding of the issues and solutions related to implementationæ
  • Require a fundamental understanding of packet telephony technologies that are common for both enterprise and service provider applications



  • Working knowledge of networking fundamentals, including LANs, WANs, and IP switching and routingæ
  • Ability to configure and operate Cisco routers and switches and to enable VLANs and DHCPæ
  • Knowledge of traditional PSTN operations and technologies

Expected Duration

5 day

Course Objectives

1.æIntroduction to Voice Gateways

  • Cisco UC Networks and the Role of Gateways
  • Gateway Call Routing and Call Legsæ
  • Gateway Voice Ports Configurationæ
  • DSP Functionality, Codecs, and Codec Complexity

2.æVoIP Call Legs

  • VoIP Call Leg Characteristicsæ
  • VoIP Media Transmissionæ
  • H.323 Signaling Protocolæ
  • SIP Signaling Protocolæ
  • MGCP Signaling Protocol
  • Requirements for VoIP Call Legsæ
  • VoIP Call Legs Configuration

3.æDial Plan Implementation

  • Call Routing and Dial Plansæ
  • Digit Manipulationæ
  • Path Selection Configuration
  • Calling Privileges Configuration

4.æGatekeeper and CUBE Implementation

  • Fundamentals of Gatekeepersæ
  • Cisco Unified Border Element


  • QoS Mechanisms and Models
  • Classification, Marking, and Link Efficiency Mechanisms
  • Managing Congestion and Rate Limiting
  • Cisco AutoQoS