SIP Advanced (Session Initiation Protocol)

Description

This is an advanced SIP training with a lot of practical exercises. Functions such as transaction handling, dialogs, different error situations, timers and a lot of other topics are covered in detail. You will learn how SIP works within IP-telephony, as well as multimedia solutions, such as presence and Instant Messaging (IM). This course will cover how SIP works both in wireline and wireless solutions and you will need basic knowledge within VoIP and SIP to participate in this course.

Target group

The target group for this course is people that need to understand the SIP signalling protocol in detail. Among these people we will find testers, developers and implementors. The course covers wired solutions as well as wireless.

Prerequisites

Basic knowledge within data communication equivalent to our course Data Communications Fundamentals, basic knowledge within within the TCP/IP protocol suite equivalent to our course TCP/IP and basic knowledge about Voice over packet networks equivalent to our course SIP Fundamentals. Knowledge Test Explore your knowledge and skills in our SIP test for free today.Our team of training advisors can make a recommendation and suggestion so it suits your specific need. SIP Test

Other

This course is available as scheduled training and the presentation is given in English or Swedish. The course is mixing theory and practical exercises. We can also give this course as on-site training. If you are interested in customized education, don’t hesitate to contact us for further information.

Agenda

Start of Day 1

Designing a VoIP Protocol

Whiteboard discussion on the generic needs and features of a generic VoIP signalling protocol, and which design SIP

The Basics

Background, history and Internet heritage
Main components; servers and clients.
Benefits and rules with different transport protocols
-UDP, TCP, SCTP.
Basic sessions and SIP mobility features
-Proxy and Redirect mode
-SIP Methods & Response codes
-SIP Registration
SDP Basics

Addressing and Registration

Message structure and format rules.
-Mandatory headers and parameters.
-Addressing and using URIs
SIP Registration
-Expiration and Deregistration
-Registration rules
-The location service

Exercise 1 – Registration and Basic Calls

Registration issues
Basic SIP sessions with SIP proxy and location server
Using Wireshark for traffic monitoring
SIP Signalling and SDP Negotiation analysis
Methods; INVITE, BYE, REGISTER

Message Forwarding

SIP message routing rules
-RequestURI and Route headers
-Via header response routing
Detailed proxy behaviour
-Location server lookup
-Request forwarding
-Response processing
-Loop detection/prevention
Using DNS for inter-domain signalling
-SIP related DNS records; NAPTR; SRV, A/AAAA
-Load sharing and redundancy features

Session Management

Establishing sessions with INVITE
Using SDP for negotiation
-SDP Offer/Answer Model
Dialog creation
-Early Dialogs and UPDATE
Dialog state management
-Routesets, Record-Routing, URI’s, and Dialog-ID.

Exercise 2 – Call Signalling Details

Inter-domain call setup and routing
DNS usage
RTP details
SIP Header analysis
Usage of Request-URI
Record-routing examples
ReINVITE’s or UPDATE for session re-negotiation

End of Day 1

Start of Day 2
Step by step walkthrough of advanced call scenario.

State handling

Statefullness in SIP Servers
-Limitations of stateless servers
-Transaction and dialog-stateful servers
-Registration stateful servers.

The transaction layer
-Client and server transaction state machines
-The bugs and fixes for Non-INVITE transactions

Protocol extensions and updates

Extending the SIP protocol
-Using OPTIONS
-Negotiation extensions
-Requiring extensions
-Handling new SIP methods in old proxies
Reliable provisional responses
-PRACK
-RSeq and RAck
Forking and Cancelling requests
-Cancel and stateful proxies
Caller preferences and Callee Capabilities
-Addressing and Registration extensions
Using Early Media
-Simplex or Full-duplex
-Issues with forking.
Quality of Service and SIP/SDP

Exercise 3 – Forking and CANCEL

Forking Calls – When and Why?
Response processing at forking.
Method: CANCEL
Parallel and sequential forking.

Security

Firewalling SIP servers and clients.
Encryption and Authentication – How?
Firewalls and NAT/PATFirewalls and NAT/PAT
- Session Traversal Utilities for NAT
-Traversal Using Relays around NAT
-SIP Outbound extension for NAT
-Symmetric Responses and Connection reuse
-Global Routable User agent URI, GRUU
SIP Privacy and Authenticity
-S/MIME examples for end to end security
-Privacy and Anonymity
Securing the media channel
- Secure RTP with ZRTP or MIKEY

Theoretical Exercise

SIP Quiz

End of Day 2

Start of Day 3

Services and Applications

Service creation possibilities with SIP
-Overview of SIP-CGI, CPL, Java servlets, Jain, OSA/Parlay
-Service examples with CPL
SIP Basic call-services and PBX-like features
-Call-forwarding, voicemail, CLIR/CLIP, etc
Call-transfer and Call-Pickup
-REFER and Replaces
3rd party call control
SIP for events
-The presence architecture
-Triggering presence
-Dialog issues
Instant messaging in SIP
-Stand-alone messages with MESSAGE
-Session based messaging with MSRP
B2BUA (Back to back User Agent)
-Requirements and Possibilities

Exercise 4 – Services

Presence
Instant messaging
Methods; SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE

Classic Telepony Using SIP

Sending DTMF in VoIP
-DTMF and RTP, rfc2833/4733
-INFO and SUBSRIBE/NOTIFY
Merging PSTN Networks and SIP
-SIP for telephones – SIP-T
-Q.1912.5– SIP-I
E.164 Phonenumbers and SIP-addresses
-Tel: and SIP: URIs
-Address translation, interworking
-DNS and ENUM
Mobile SIP Telephony
-IMS, IP Multimedia Subsystem

Exercise 5 – Security and DNS