Price: $1,699.00

Course Number: 627
Length: 2 Days
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Advanced SIP Training by TONEX is a more technical SIP course.

Advanced SIP training course provides a technical details of SIP protocol. Advanced Session Initiation Protocol (SIP) training course gives you the solid technical details you need to architect, design, implement, verify, troubleshoot and maintain SIP in your application, regardless of vendor.

SIP is a text-based protocol, similar to HTTP and SMTP, for initiating interactive communication sessions between users covering services such as voice, video, chat, interactive games, and virtual reality.
Learn about:

  • Architecture of SIP
  • User agents, presence agents, gateways, and servers (proxy, stateful, stateless, call stateful, redirect, forking, and session border controller (SBC)
  • SIP Security
  • SIP methods
  • Advanced call flows
  • SIP in IMS and Voice over LTE

Who Should Attend

Software/System/Hardware engineers and developers


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 the role of SIP in IMS, HSPA/HSPA+, LTE
  • Examine SIP Other Important Topics


VoIP Protocol Stack

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

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

Overview of 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


  • 3GPP SIP Decisions
  • SIP from terminal to network
  • SIP between network call nodes
  • SIP Protocol & Service analysis
  • Deployments in 3GPP
  • 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


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