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Course 627: Advanced SIP

Course #: 627
Course Type: On-site & Public
Duration: 2 days
Price: $1699
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Available Training Formats
On-site Public
Computer Based Training Online

Description

The Advanced SIP course provides a technical details on SIP.

 

 

 

Objectives

After successfully completing the course the attendees will:

  • Understand advanced topics of SIP
  • Understand advanced SDP concepts
  • Explore the architect and components of SIP
  • Behavior of SIP Clients and Servers
  • Examine behavior of SIP User Agents
  • Examine behavior of SIP Proxy and Redirect Servers
  • Analyze security considerations for SIP
  • Understand SIP-T and SIP interworking with ISUP networks (PSTN)
  • Understand IP Multimedia (IMS) Using SIP in UMTS and CDMA2000
  • Understand the role of SIP & IMS in 3G
  • Examine SIP Other Important Topics

Course Outline

Advanced SIP Training content

VoIP Protocol Stack

  • RTP and RTCP
  • H323 Protocols
  • Session Initiation Protocol (SIP)
  • Media Gateway Control Protocol (MGCP)
  • Megaco/H248

Session Description Protocol (SDP)

  • SDP Fields and Parameters
  • SDP Media Information
  • Type of media (video, audio, etc)
  • The transport protocol (RTP/UDP/IP, H320, etc)
  • The format of the media (H261 video, MPEG video, etc)
  • Multicast address for media
  • Transport Port for media

Session Initiation Protocol (SIP)

  • Context of SIP
  • SIP Functions
  • SIP Hardware Architecture and User Agent (UA)
  • SIP Servers
  • SIP Protocol Model and Addressing
  • Request and Response Model
  • Locating a SIP Server
  • SIP Transaction
  • SIP Invitation
  • Locating a User
  • Changing an Existing Session
  • Registration Services
  • SIP Message Format
  • SIP Methods
  • Status Code Definitions
  • Analysis of SIP Response Codes
  • SIP Header Fields
  • SIP Registration
  • The Registrar
  • SIP Registration with Authentication
  • User Cancels Registration
  • SIP Call Setup via Proxy Server

Behavior of SIP Clients and Servers

  • Requests and Responses
  • Source Addresses, Destination Addresses and Connections
  • Unicast and Multicast UDP
  • Reliability for BYE, CANCEL, OPTIONS, REGISTER
  • Reliability for INVITE Requests
  • Reliability for ACK Requests
  • ICMP Handling

Behavior of SIP User Agents

  • Caller Issues Initial INVITE Request
  • Callee Issues Response
  • Caller Receives Response to Initial Request
  • Caller or Callee Generate Subsequent Requests
  • Receiving Subsequent Requests

Behavior of SIP Proxy and Redirect Servers

  • Redirect Server
  • User Agent Server
  • Proxy Server
  • Proxying Requests
  • Proxying Responses
  • Stateless Proxy: Proxying Responses
  • Stateful Proxy: Receiving Requests
  • Stateful Proxy: Receiving ACKs
  • Stateful Proxy: Receiving Responses
  • Stateless, Non-Forking Proxy
  • Forking Proxy

Security Considerations for SIP

  • Confidentiality and Privacy: Encryption
  • End-to-End Encryption
  • Privacy of SIP Responses
  • Encryption by Proxies
  • Hop-by-Hop Encryption
  • Via field encryption
  • Message Integrity and Access Control
  • Authentication
  • Trusting networks and responses
  • SIP Authentication using HTTP Basic and Digest Schemes
  • Management Information Base for SIP
  • The Stream Control Transmission Protocol (SCTP) as a Transport for for the SIP
  • Compressing the SIP

SIP-T: SIP Interworking with ISUP Networks

  • PSTN Interwork with SIP networks
  • ISUP & SIP Signaling
  • Routing of E164 numbers
  • Mapping between SIP & ISUP parameters
  • Handling of Telephony Events such as DTMF and Ringback
  • SIP Messages, SIP- vs TEL-URIs, SIP Proxies, SIP/ISUP Gateways, ENUM DNS, IAM-INVITE

SIP & IMS in 3G: IP Multimedia Using SIP in UMTS

  • 3GPP SIP Decisions
  • SIP from terminal to network
  • SIP between network call nodes
  • SIP Protocol & Service analysis
  • Deployments in UMTS and CDMA2000 Networks
  • 3GPP IMS
  • Security
  • Charging
  • QoS and Signaling
  • IMS Architecture : P-CSCF, I-CSCF, and S-CSCF
  • IMS Building Blocks
  • Applications Layer
  • End-user telephony service logic
  • AIN call trigger points
  • Non-telephony based services
  • APIs for enterprise & legacy applications
  • Session Control Layer
  • End Point Registration
  • Session setup
  • QoS establishment
  • Transport & Endpoint Layer
  • Bearer Services, Media Conversion (PCM > IP)
  • Special functions: announcements, touch tones collection, voice recognition, speech
  • Signaling Flows
  • Instant Messaging & Presence
  • Push-to Talk
  • Conferencing
  • OSA and Parlay


SIP Other Important Topics

  • A SIP Event Notification Extension for Resource Lists
  • The Message Session Relay Protocol
  • The Extensible Markup Language (XML) Configuration Access Protocol (XCAP)
  • RPID: Rich Presence: Extensions to the Presence Information Data Format (PIDF)
  • CIPID: Contact Information in Presence Information Data Format
  • A Mechanism for Content Indirection in SIP Messages
  • Enhancements for Authenticated Identity Management in the Session Initiation Protocol Globally Routable User Agent (UA) URIs (GRUU) in the SIP
  • DHCP and DHCPv6 Option for SIP Servers
  • HTTP Digest Authentication Using Authentication and Key Agreement (AKA)
  • Integration of Resource Management and SIP
  • SIP Extension for Instant Messaging
  • S/MIME AES Requirement for SIP
  • Advanced Instant Messaging Requirements for the SIP
  • Transcoding Services Invocation in the SDP Using 3Pcc
  • E.164 numbers with the SIP
  • SIP Interoperability Testing

Who Should Attend

Software/System/Hardware engineers and developers

Prerequisites

SIP fundamentals course or equivalent training program is recommended.

 

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