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Course Outline

Learning Objectives:

  • Explain call signaling and setup procedures in voice networks.
  • Describe voice media transport and bandwidth requirements for VoIP calls.
  • Detail SIP standards, services, messages, and return codes.
  • Outline basic call setup using SIP.
  • Analyze SIP flows and SDP.
  • Explain the registration process and making calls via a SIP Server.
  • Discuss IP PBX functionality and call conferencing.
  • Clarify SRV records and DNS roles.
  • Explain uri/url/urn, ENUM, and NAPTR records.
  • Describe service-to-address mapping.
  • Detail SIP-T and SIP-I protocols.
  • Explain SIP early media and SIP trunks.
  • Analyze call flows between PSTN and IP networks using SIP.
  • Describe Secure SIP, Secure RTP, and Secure RTCP.
  • Review typical Secure SIP implementations.

Practical Labs:

  • Lab 1: Practical SIP implementation in the LAN using Xlite
  • Lab 2: Analyzing SIP packets with Wireshark
  • Lab 3: SDP, Presence, and Instant Messaging
  • Lab 4: Call flows with a SIP Server
  • Lab 5a: SIP registration with DNS
  • Lab 5b: Call flows with DNS
  • Lab 6: SIP trunks
  • Lab 7: Security implementation with IPSec
  • Lab 8: Security implementation with Secure SIP

Requirements

  • Delegates are expected to possess a working knowledge of TCP/IP.
  • A foundational understanding of VoIP is recommended.
 21 Hours

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