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SIP

The Universal Signaling Protocol

ExperTeach Networking Logo
In the meantime, the Session Initiation Protocol (SIP) has established itself as the most important signaling protocol both in the enterprise environment and in the provider network. One major advantage of SIP is that it can be easily extended: New formats are no problem, both synchronous and asynchronous data streams can be initiated, and the communications partners can have a peer-to-peer or client-server relationship. After attending this course, the students will be familiar with the benefits, special features, and application options of the SIP architecture in general and of SIP trunking in particular.

Course Contents

  • The Components SIP Proxy, Location Server, and User Agent
  • Back-to-Back User Agent (B2BUA) and Session Border Controller (SBC)
  • SIP Protocol: Message Types and their Setup
  • Typical SIP Processes during Connection Setup and a SIP Call
  • SIP URIs and Tel URIs: Address Formats, Identities, and their Application
  • SDP: Setup, Options, and Profiles
  • VoIP and Video over IP (RTP and Signaling) Data Streams
  • Features—Instant Messaging—Presence
  • Interaction of SIP with NAT and Firewalls
  • Fax with T.38 and Interaction with SIP
  • SIP as a Protocol in the IP Multimedia Subsystem (IMS)
  • Application of SIP in Provider Networks
  • SIP Trunking

Practice-related presentations and the analysis of traces will make the course contents easier to understand.

Print E-Book PDF Symbol You will receive the comprehensive documentation package of the ExperTeach Networking series – printed documentation, e-book, and personalized PDF! As online participant, you will receive the e-book and the personalized PDF.

Target Group

Network designers belong to the target group of the course as much as employees who have to be familiar with SIP on the protocol layer.

Knowledge Prerequisites

A well-based know-how of the voice and IP sectors is the prerequisite for this course. Basic VoIP know-how will be very helpful for course participants.

Alternatives

If you are interested in a general overview of VoIP including SIP, the VoIP Fundamentals—SIP, H.323 & Co. in Application course is the right choice for you.

 

1 Fields of Application of SIP
1.1 The Basic Idea
1.1.1 Signaling in General
1.2 Payload Data Transport
1.2.1 RTP—Transport and Reconstruction Function
1.2.2 RTCP—Information on RTP Connections
1.2.3 Messaging
1.2.4 End Devices
1.3 SIP in the Enterprise Environment
1.4 SIP in the Provider Environment
1.5 SIP and WebRTC
   
2 SIP—The Basics
2.1 SIP—Session Initiation Protocol
2.1.1 Classification in the ISO/OSI Model
2.2 The Components of the SIP Architecture and their Tasks
2.2.1 The End Devices: User Agents
2.2.2 The Gateways
2.2.3 The SIP Proxy
2.3 Protocol Setup
2.3.1 SIP Requests—SIP Methods
2.3.2 Responses from 100 Trying to 600 Busy Everywhere
2.4 A Session in Progress
2.4.1 A session is not set up (1)
2.4.2 A session is not set up (2)
   
3 SIP Advanced
3.1 The SIP Message
3.2 Registration and Control
3.2.1 Registration of a SIP UA
3.2.2 Proxy Authentication
3.3 A Session in Detail
3.3.1 INVITE
3.3.2 100 Trying
3.3.3 180 Ringing
3.3.4 200 OK Responding to the INVITE
3.3.5 ACK for the INVITE
3.3.6 Disconnection and BYE—Important Details
3.3.7 Call Forking
3.4 Result Control and SIP
3.4.1 Events
3.4.2 SUBSCRIBE and NOTIFY
3.5 SIMPLE
3.5.1 PUBLISH
3.5.2 Instant Messaging and MESSAGE
3.6 Further Request Types
3.6.1 OPTIONS
3.6.2 PRACK—Reliable Acknowledgment
3.6.3 UPDATE
3.6.4 REFER
3.7 Application in Provider and Enterprise Structures
3.8 Session Description Protocol
3.8.1 Setup of the Message Body with SDP
3.8.2 SDP for Advanced Users
3.8.3 RTP Profiles
3.9 Dial Tones—DTMF
3.10 Classic Features
3.10.1 Call Hold and Consultation Hold
3.10.2 Music On Hold
3.10.3 Call Forwarding (Unconditional)
3.10.4 Call Transfer (Unattended)
3.10.5 3-Party Conference Call
3.10.6 Completion of Call to Busy Subscriber
3.10.7 Features for VoIP vs. PSTN
3.10.8 RFC 3842: Voice Mailboxes
3.10.9 RFC 3680: Registrations
   
4 SIP in Network Operation
4.1 Security Aspects
4.1.1 VoIP and Stateful Firewalls
4.1.2 Encryption: SIPS and SRTP
4.2 Tools to Handle NAT
4.2.1 IADs and ALGs
4.2.2 STUN
4.2.3 Interactive Connection Establishment (ICE)
4.3 Session Border Controller
4.4 The IMS—Core Structure of the NGN
4.4.1 The IMS Architecture
4.4.2 Signaling in the IMS—The Components
4.4.3 P-Header Extensions
4.5 Fax Solutions
4.5.1 Fax as a Default VoIP Call
4.5.2 Fax Transmission with T.38
4.6 SIP Trunking
4.6.1 SIP Trunking Architecture and Security Aspects
4.6.2 Registration Mode
4.6.3 Static Mode
4.6.4 Enterprise Public Identity
4.6.5 Handling Calls
4.6.6 Media Endpoints
4.6.7 Emergency Calls
4.7 Fault Tolerance and Load-balancing
4.8 Overload Control
4.8.1 Causes of Overload
4.8.2 Previous SIP Mechanisms
4.8.3 Via Header Extension
4.8.4 Load Control Event Package
   
A SIP Response Codes
A.1 Response Codes
   
B List of Abbreviations

Classroom training

Do you prefer the classic training method? A course in one of our Training Centers, with a competent trainer and the direct exchange between all course participants? Then you should book one of our classroom training dates!

Hybrid training

Hybrid training means that online participants can additionally attend a classroom course. The dynamics of a real seminar are maintained, and the online participants are able to benefit from that. Online participants of a hybrid course use a collaboration platform, such as WebEx Training Center or Saba Meeting. To do this, a PC with browser and Internet access is required, as well as a headset and ideally a Web cam. In the seminar room, we use specially developed and customized audio- and video-technologies. This makes sure that the communication between all persons involved works in a convenient and fault-free way.

Online training

You wish to attend a course in online mode? We offer you online course dates for this course topic. To attend these seminars, you need to have a PC with Internet access (minimum data rate 1Mbps), a headset when working via VoIP and optionally a camera. For further information and technical recommendations, please refer to.

Tailor-made courses

You need a special course for your team? In addition to our standard offer, we will also support you in creating your customized courses, which precisely meet your individual demands. We will be glad to consult you and create an individual offer for you.
Request for customized courses
PDF SymbolYou can find the complete description of this course with dates and prices ready for download at as PDF.

In the meantime, the Session Initiation Protocol (SIP) has established itself as the most important signaling protocol both in the enterprise environment and in the provider network. One major advantage of SIP is that it can be easily extended: New formats are no problem, both synchronous and asynchronous data streams can be initiated, and the communications partners can have a peer-to-peer or client-server relationship. After attending this course, the students will be familiar with the benefits, special features, and application options of the SIP architecture in general and of SIP trunking in particular.

Course Contents

  • The Components SIP Proxy, Location Server, and User Agent
  • Back-to-Back User Agent (B2BUA) and Session Border Controller (SBC)
  • SIP Protocol: Message Types and their Setup
  • Typical SIP Processes during Connection Setup and a SIP Call
  • SIP URIs and Tel URIs: Address Formats, Identities, and their Application
  • SDP: Setup, Options, and Profiles
  • VoIP and Video over IP (RTP and Signaling) Data Streams
  • Features—Instant Messaging—Presence
  • Interaction of SIP with NAT and Firewalls
  • Fax with T.38 and Interaction with SIP
  • SIP as a Protocol in the IP Multimedia Subsystem (IMS)
  • Application of SIP in Provider Networks
  • SIP Trunking

Practice-related presentations and the analysis of traces will make the course contents easier to understand.

Print E-Book PDF Symbol You will receive the comprehensive documentation package of the ExperTeach Networking series – printed documentation, e-book, and personalized PDF! As online participant, you will receive the e-book and the personalized PDF.

Target Group

Network designers belong to the target group of the course as much as employees who have to be familiar with SIP on the protocol layer.

Knowledge Prerequisites

A well-based know-how of the voice and IP sectors is the prerequisite for this course. Basic VoIP know-how will be very helpful for course participants.

Alternatives

If you are interested in a general overview of VoIP including SIP, the VoIP Fundamentals—SIP, H.323 & Co. in Application course is the right choice for you.

 

1 Fields of Application of SIP
1.1 The Basic Idea
1.1.1 Signaling in General
1.2 Payload Data Transport
1.2.1 RTP—Transport and Reconstruction Function
1.2.2 RTCP—Information on RTP Connections
1.2.3 Messaging
1.2.4 End Devices
1.3 SIP in the Enterprise Environment
1.4 SIP in the Provider Environment
1.5 SIP and WebRTC
   
2 SIP—The Basics
2.1 SIP—Session Initiation Protocol
2.1.1 Classification in the ISO/OSI Model
2.2 The Components of the SIP Architecture and their Tasks
2.2.1 The End Devices: User Agents
2.2.2 The Gateways
2.2.3 The SIP Proxy
2.3 Protocol Setup
2.3.1 SIP Requests—SIP Methods
2.3.2 Responses from 100 Trying to 600 Busy Everywhere
2.4 A Session in Progress
2.4.1 A session is not set up (1)
2.4.2 A session is not set up (2)
   
3 SIP Advanced
3.1 The SIP Message
3.2 Registration and Control
3.2.1 Registration of a SIP UA
3.2.2 Proxy Authentication
3.3 A Session in Detail
3.3.1 INVITE
3.3.2 100 Trying
3.3.3 180 Ringing
3.3.4 200 OK Responding to the INVITE
3.3.5 ACK for the INVITE
3.3.6 Disconnection and BYE—Important Details
3.3.7 Call Forking
3.4 Result Control and SIP
3.4.1 Events
3.4.2 SUBSCRIBE and NOTIFY
3.5 SIMPLE
3.5.1 PUBLISH
3.5.2 Instant Messaging and MESSAGE
3.6 Further Request Types
3.6.1 OPTIONS
3.6.2 PRACK—Reliable Acknowledgment
3.6.3 UPDATE
3.6.4 REFER
3.7 Application in Provider and Enterprise Structures
3.8 Session Description Protocol
3.8.1 Setup of the Message Body with SDP
3.8.2 SDP for Advanced Users
3.8.3 RTP Profiles
3.9 Dial Tones—DTMF
3.10 Classic Features
3.10.1 Call Hold and Consultation Hold
3.10.2 Music On Hold
3.10.3 Call Forwarding (Unconditional)
3.10.4 Call Transfer (Unattended)
3.10.5 3-Party Conference Call
3.10.6 Completion of Call to Busy Subscriber
3.10.7 Features for VoIP vs. PSTN
3.10.8 RFC 3842: Voice Mailboxes
3.10.9 RFC 3680: Registrations
   
4 SIP in Network Operation
4.1 Security Aspects
4.1.1 VoIP and Stateful Firewalls
4.1.2 Encryption: SIPS and SRTP
4.2 Tools to Handle NAT
4.2.1 IADs and ALGs
4.2.2 STUN
4.2.3 Interactive Connection Establishment (ICE)
4.3 Session Border Controller
4.4 The IMS—Core Structure of the NGN
4.4.1 The IMS Architecture
4.4.2 Signaling in the IMS—The Components
4.4.3 P-Header Extensions
4.5 Fax Solutions
4.5.1 Fax as a Default VoIP Call
4.5.2 Fax Transmission with T.38
4.6 SIP Trunking
4.6.1 SIP Trunking Architecture and Security Aspects
4.6.2 Registration Mode
4.6.3 Static Mode
4.6.4 Enterprise Public Identity
4.6.5 Handling Calls
4.6.6 Media Endpoints
4.6.7 Emergency Calls
4.7 Fault Tolerance and Load-balancing
4.8 Overload Control
4.8.1 Causes of Overload
4.8.2 Previous SIP Mechanisms
4.8.3 Via Header Extension
4.8.4 Load Control Event Package
   
A SIP Response Codes
A.1 Response Codes
   
B List of Abbreviations

Classroom training

Do you prefer the classic training method? A course in one of our Training Centers, with a competent trainer and the direct exchange between all course participants? Then you should book one of our classroom training dates!

Hybrid training

Hybrid training means that online participants can additionally attend a classroom course. The dynamics of a real seminar are maintained, and the online participants are able to benefit from that. Online participants of a hybrid course use a collaboration platform, such as WebEx Training Center or Saba Meeting. To do this, a PC with browser and Internet access is required, as well as a headset and ideally a Web cam. In the seminar room, we use specially developed and customized audio- and video-technologies. This makes sure that the communication between all persons involved works in a convenient and fault-free way.

Online training

You wish to attend a course in online mode? We offer you online course dates for this course topic. To attend these seminars, you need to have a PC with Internet access (minimum data rate 1Mbps), a headset when working via VoIP and optionally a camera. For further information and technical recommendations, please refer to.

Tailor-made courses

You need a special course for your team? In addition to our standard offer, we will also support you in creating your customized courses, which precisely meet your individual demands. We will be glad to consult you and create an individual offer for you.
Request for customized courses

PDF SymbolYou can find the complete description of this course with dates and prices ready for download at as PDF.