This course is well suited to Network Planners, Designers, and Engineers requiring an understanding of SIP.
It is assumed that delegates will have a working knowledge of TCP/IP. Additionally, a basic understanding of VoIP would be beneficial.
3 days. Hands on.
This course is available on site only. Please call for details.
This three 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 PBXs (e.g., Trixbox).
Key objectives include:
- 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 SIP in the LAN with Xlite
Examine SIP Packets using Wireshark
SDP, Presence and IM
Call Flows with SIP Server
SIP Registration with DNS
Call Flows with DNS
Security with IPSec
Security with Secure SIP
Practical sessions use Cisco 1700 or 2600 or 2800 series voice capable routers and Cisco Firewalls. IP Telephones, softphones, analogue telephones and Trixbox/Asterisk servers as required.