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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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