Overview
The 3 day practical course covers the messages and call flows of the Session Initiation Protocol (SIP) and its use in voice networks. This course is a mixture of theory and practice (utilising protocol analyser traces where appropriate for explanation and troubleshooting) with practical VoIP configured using IP telephones, softphones, voice capable Cisco routers and SIP IP PBX,s (e.g. Trixbox).
Audience:
Network Planners, Designers, and Engineers requiring an understanding of SIP.
The course is around 40% practical.
Requirements
- It is assumed that delegates will have a working knowledge of TCP/IP.
- A basic understanding of VoIP would be beneficial.
Course Outline
Objectives:
- Describe call signalling and setup in the voice network
- Describe carrying of voice media and bandwidth requirements for VoIP calls
- Describe SIP standards, services, messages and return codes
- Describe basic call setup using SIP
- Describe SIP flows and SDP
- Describe registration process and making calls with a SIP Server
- Describe IP PBX and Call Conferences
- Describe SRV records and DNS
- Describe uri/url/urn, ENUM and NAPTR Records
- Describe mapping of services to an address
- Describe SIP-T and SIP-I
- Describe SIP early media and SIP trunks
- Describe call flows between PSTN and IP using SIP
- Describe Secure SIP, Secure RTP and Secure RTCP
- Describe typical Secure SIP implementations
Practical Exercises:
- Lab Exercise 1: Practical SIP in the LAN with Xlite
- Lab Exercise 2: Examine SIP Packets using Wireshark
- Lab Exercise 3: SDP, Presence and IM
- Lab Exercise 4: Call Flows with SIP Server
- Lab Exercise 5a: SIP Registration with DNS
- Lab Exercise 5b: Call Flows with DNS
- Lab Exercise 6: SIP Trunks
- Lab Exercise 7: Security with IPSec
- Lab Exercise 8: Security with Secure SIP