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SIP
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| Course Name: |
SIP Training: Session Initiation Protocol (SIP) Training |
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| Deployment Options: |
Onsite - Instructor-Led Training |
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| Course Duration: |
3-4 days depending on audience background and options |
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| Related Courses |
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| Introduction: |
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SIP, the Session Initiation Protocol, is a signaling protocol for conferencing, telephony, presence,
events notification and instant messaging.
It is an application-layer control (signaling) protocol for creating, modifying and terminating sessions with one
or more participants. These sessions include Internet multimedia conferences, Internet telephone calls and multimedia
distribution. Members in a session can communicate via multicast or via a mesh of unicast relations, or a combination
of these. Session Initiation Protocol (SIP) builds on the IP communications foundation by providing a standards-based
approach to enabling IP communications with numerous devices and applications.
SIP invitations used to create sessions carry session descriptions which allow participants to agree on a set of
compatible media types. SIP supports user mobility by proxying and redirecting requests to the user's current location.
Users can register their current location. SIP is not tied to any particular conference control protocol. SIP is
designed to be independent of the lower-layer transport protocol and can be extended with additional capabilities.
Session controllers promise to enable the same ubiquity, quality, and security for VoIP that the PSTN offers today,
only in the more flexible, efficient, and economical manner that IP makes possible.
The SIP fundamentals course provides an overview of SIP, its components, and how it works. It covers data networking
principles to telco engineers and signaling principles to IP engineers. It also outlines SIP implementations on
the market in the form of single-line gateways, proxy servers, media gateways, Java toolkits, encoders/decoders
and session authenticators. |
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| Audience: |
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| Individuals who wish to develop a basic knowledge of SIP. Essential course for anyone involved
in the development and operation of voice or data networks, wireless communications protocol, mobility technologies,
and instant messaging. |
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| Prerequisites: |
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| This is an introductory course with no prerequisites. |
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| Customize it: |
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| This 3-4-day course will be customized to your needs and specifications. Eno.com will assist
you in identifying those needs and specifications. A word to the wise, there are many vendors of wireless training.
They will typically have a broad and general course, one size fits all, already developed and just put your organization?s
name on the title slide. This minimizes their effort and time investment. At Eno.com, every course is made to your
exact and exacting specifications. We help you ensure what you are getting is what you really need even if at the
beginning you weren't too sure of what that was. We fit the class to your needs. We never fit you into our standard,
one size fits all, class. |
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| Objectives: |
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After successfully completing the course the attendees will:
- Understand basics of VoIP
- Explore Where, why, and how SIP is used
- Comprehend the basics of SIP
- Understand the architect and components of SIP
- Understand the differences between SIP and H.323
- Understand H.323-SIP-SS7 Interworking
- Review SIP-T concept and architecture
- Understand how to size up and choose from available SIP products
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| Course Outline |
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Executive Summary
- Circuit-switched network signaling
- Introduction to SS7
- SS7 signaling
- Operation of voice and data networks
- VoIP Basics
- IP Signaling protocols
- Initiating, managing and terminating voice and video sessions
- Set up, modify, and tear down multimedia sessions over the Internet
- The session initiation protocol (SIP)
- A new signaling protocol
- Key services through the use of SIP
- SIP capabilities
- SS7-SIP interworking requirements
SIP Overview
- Fundamentals of how SIP works
- SIP context and architectures
- SIP sessions
- SIP flows
- Core SIP
- Encapsulation
- Translation
- SIP content negotiation
- Session description protocol (SDP)
- Security considerations
- HTTP and SMTP, SIP
- SIP extended features and services
- Call control services, mobility, interoperability with existing telephony systems
- Standardization status
- Supported services
- Proprietary extension and negotiation mechanisms
- Interoperability of services and features
- Interworking with PSTN
- Service creation issues
- Basic call features
- Quality of Service issues
- Network services
- Conferencing and addressing
- SIP, H.323, MGCP and Megaco
- Basic SIP Communication Services
- Integrating SIP with PSTN
- SIP in IPv4 and IPv6
- SIP and 3G Wireless
- Session Initiation Protocol for Telephones (SIP-T)
- SIP and SIGTRAN
SIP System Operations
- SIP Parameters
- Protocols
- User Agents
- Call Processors
- Customer Status
- Address Tracking
- Call Forwarding
SIP Protocol Operation
- Client/Server transactions
- Proxy servers
- SIP messages
- Transport layer
- Extending SIP
- Extension negotiation
- Technical details of SIP extensions
- SIP Extensions
- Session Description Protocol (SDP)
- SDP packets
- SIP timer
- SIP programming
- JAIN API
- SIP Lite
- SIP servlets
- SIP for J2ME
- SIP and SOAP
- SIP and VoiceXML
SIP Entities
- Components of SIP
- SIP Clients
- SIP as a peer-to-peer protocol
- User Agents (UAs) as the peers in a session
- User agent client (UAC)
- User agent server (UAS)
- SIP Servers
- Using A Proxy Server
- Using a Redirect Server
- Proxy Server
- Redirect Server
- Registrar
SIP Messages
- Message Types
- Message Parts
- Message Samples
- Requests
- Responses
- Header Fields
- Bodies
- Framing SIP Messages
- Status Code Definitions
- Informational 1xx
- Successful 2xx
- Redirection 3xx
- Request Failure 4xx
- Server Failure 5xx
- Global Failures 6xx
SIP Parameters
- Header Fields
- Option Tags
- Warning Codes (warn-codes)
- Methods and Response Codes
- Reason Protocols
- Security Mechanism Names
- Compression Schemes
SIP vs. H.323
- Robustness
- Security
- Legacy
- Political Issues
- Status Update
- References
SIP-T
- SIP-T for ISUP-SIP interconnections
- SIP-T flows
- SIP bridging (PSTN - IP - PSTN)
- PSTN origination - IP termination
- IP origination - PSTN termination
- SIP-T roles and behavior
- Components of the SIP-T protocol
- Support for mid-call signaling
SIP Signaling Flows in 3G/UMTS
- UMTS core network architecture
- Call management in UMTS R4/R5
- Soft handover
- Hard handover
- Registration
- Session initiation
- Session termination
- Roaming scenarios
SIP Products and Trends
- Application Servers
- Applications
- Session Border Controllers
- SIP based Services
- SIP Gateways
- SIP Hardware Appliances
- SIP Phones
- SIP Presence and Messaging Servers
- Software Development Kits
- Testing and Simulation
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| Other Expertise: |
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