Powerful 5-Day Hands on Course

Course Description
This course is a detailed review of the SIP protocol. The emphasis is on the protocol itself, not any specific application of SIP. Hands on training is used at least 50% of the course. This course is very popular and the course reviews are "raving".
Intended Audience
This course is intended for technical personnel that need a vendor neutral understanding of the SIP protocol.
Prerequisites
Student prerequisites are as follows: TCP/IP, VoIP, and Gatekeeper knowledge and understanding.
Course Outline
- Why does IMS use SIP?
- Circuit vs. packet concepts
- Circuit-based bearer channel vs. Packet-based bearer channel (DS0 vs. RTP)
- SS7/ISDN vs. SIP, MEGACO
- The IMS universal network design goal – build one horizontal network that can support all mobile services from one network
- VoIP Protocols comparison
- Why SIP?
- A short history of SIP
- IP Routing and Switching Overview
- IP Routing
- Ethernet Essentials
- TCP and UDP Essentials
- How VoIP uses TCP
- How VoIP uses UDP
- IMS Architecture
- IMS and SIP
- RTP Introduction
- 3GPP/3GPP2, IETF, OMA (Open Mobile Alliance)
- Applications: Push to Talk, Instant Messaging, Presence, Telephony, Gaming
- Proxy-CSCF
- Serving-CSCF
- Interrogating-CSCF
- Policy Decision Function
- HSS
- SIP Architecture
- The SIP architecture
- UA, Proxy, Redirect, Forking,
- Multimedia Architecture
- Methods
- REGISTER
- INVITE and ACK
- UPDATE
- OPTIONS
- CANCEL
- REFER
- SUBSCRIBE and NOTIFY
- MESSAGE
- BYE
- SIP responses
- 1xx Informational
- 2xx Final
- 3xx Redirection
- 4xx Client Error
- 5xx Server Error
- 6xx Global Failure
- SIP Uniform Resource Indicators (URIs)
- Generic URI information (RFC 2396)
- Direct or Proxy
- PSTN number (RFC 2808)
- Instant messaging
- Presence
- In registrations
- SIP Headers
- Via
- Branch
- Max-Forwards
- Dialog (To, From, and tag= fields)
- CSeq
- Call-ID
- Contact
- SIP reliability
- Expires
- Authentication
- IMS Signaling compression (SigComp architecture)
- Session Description Protocol (SDP)
- Learn how SIP uses SDP to define technical parameters that support a voice over IP media channel.
- Session parameters
- SDP format
- Extending SDP
- SDPng
- Media negotiation
- Changing session parameters
- IMS Related IP Services
- DHCP
- ENUM
- DNS NAPTR records
- DNS SRV records
- Regular Expressions
- DDDS algorithm
- SIP Call Flow Examples
- Call attempt – unsuccessful
- Presence subscription
- Registration
- Registration with Authentication
- Presence notification
- Instant Message Exchange
- Call setup – successful
- Call hold
- Call Forward
- Call transfer
- Unified messaging
- IMS Reference Points
- Gm, Mw, Cx, Dx, Sh, Si, Dh, Mm, Mg, Mi, Mj, Mk, Ut, Mr, Mp, Go, Gq
- IMS Call Processing
- Registration
- Service discovery
- Identity modules
- IMS Authentication and key agreement (User Identities)
- Network domain security
- Secure HTTP-based service
- Controlling bearer traffic
- Controlling the Media
- Anonymous calling (Hide Caller-ID)
- S-CSCF assignment processes
- MEGACO and SIP controlled PSTN connectivity
- Creation of via-path for response routing
- Response merging
- Record route
- Control models
- Third party call control
- Conferencing (REFER)
- Access and location information
- IMS Messaging (Immediate, Session-based, Deferred)
- RTP and RTCP (Real-Time Control Protocol)
- Dealing Packet Loss, Latency, Jitter
- How RTP defines the session
- Session Description Protocol
- The RTP profile
- The RTP payload type field
- RTP telephony events (RFC 2833)
- How RTP removes jitter
- How RTP handles packet loss
- How RTP identifies the talking party
- How RTP handles silence suppression
- How RTP handles fixed length packets (padding)
- How RTP is used to mix voice (conference calls)
- The RTP header
- RFC 2833 protocol
- RTP Control Protocol (RTCP)
- SDES
- Sender/receiver reports
- Bye reports
- IMS Presence
- SIMPLE - SIP for Instant Messaging and Presence Leveraging Extensions
- Terminology
- Framework
- Resource List Manipulation Requirements
- Authorization Policy Manipulation
- Acceptance Policy Requirements
- Notification Requirements
- Content Requirements
- General Requirements
- SIP Timers
- IMS Security
- Security for call setup
- Authentication
- S/MIME
- TLS
- Privacy and identity
- Diameter
Course Labs
- Configure PCs and Build an IP network
- Learn how to use Ethereal
- Configure the SIPURA ATA
- Configure X-Lite SIP Client
- Model the CSCF functions using the ONDO SIP Proxy
- asterisk@home
- Perform Call traces with ethereal and SIP UAs
- SIP REGISTER without authentication
- SIP REGISTER with authentication
- Simple SIP Call without INVITE authentication
- SIP call with INVITE authentication
- 100rel (PRACK)
- Busy call
- Vacant Number
- Abandoned Call
- DTMF - SIP INFO
- DTMF - RFC 2833
- DTMF – in band
- SIP NOTIFY (voice mail indication example)
- Call Forward Immediate
- Call Forward No Answer
- Call Transfer (REFER)
- Bye message with RTCP call information
- SIP Timers effect on call processing
- Model the IMS using multiple layers of proxies
|