Maziar Shoaie

M.Sc.
Senior Technical Trainer

SIP Advanced - 3 days

CONTENTS:
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 AUDIENCE:
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 implementers. 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 the TCP/IP protocol suite and about Voice over packet networks equivalent to our course SIP
Fundamentals.

EXCERCISES:
This is a theoretical training with practical exercises.

NOTE: Participants need to have their Laptop with Wireshark installed to do the exercices!

Designing a VoIP Protocol Message Forwarding

 Whiteboard discussion on the generic needs and  SIP message routing rules
features of a VoIP signaling protocol, and which - Request-URI and Route headers
choices were made when designing the SIP protocol. - Via header response routing
 Detailed proxy behavior
The Basics - Location server lookup
 Background, history and Internet heritage - Request forwarding
 Main components; servers and clients. - Response processing
- Loop detection/prevention
 Benefits and rules with different transport protocols
 Using DNS for inter-domain signaling
- UDP, TCP, SCTP.
- SIP related DNS records; NAPTR; SRV,
 Basic sessions and SIP mobility features A/AAAA
- Proxy and Redirect mode - Load sharing and redundancy features
- SIP Methods & Response codes
- SIP Registration
Session Management
 SDP Basics
 Establishing sessions with INVITE
 Different flavours of SIP, SIP-T, SIP-I, …
 Using SDP for negotiation
 SIP Truncks
- SDP Offer/Answer Model
 Dialog creation
Addressing and Registration
- Early Dialogs and UPDATE
 Message structure and format rules.
 Dialog state management
- Mandatory headers and parameters.
- Addressing and using URIs - Route-set, Record-Routing, URI’s, and Dialog-
ID.
 SIP Registration
- Expiration and Deregistration
Exercise 2 – Call Signalling Details
- Registration rules
- The location service  Inter-domain call setup and routing
 DNS usage
Exercise 1 – Registration and Basic Calls  RTP details
 Registration issues  SIP Header analysis
 Basic SIP sessions with SIP proxy and location  Usage of Request-URI
server  Record-routing examples
 Using Wireshark for traffic monitoring  ReINVITE’s or UPDATE for session re-negotiation
 SIP Signalling and SDP Negotiation analysis
 Methods; INVITE, BYE, REGISTER Step by step walkthrough of advanced call scenario.

2 Spinnaker House, Juniper Drive, SW18 1FR London, U.K. T +46 768 633 899 E maziar.shoaie@gmail.com
 Maziar Shoaie
M.Sc.
Senior Technical Trainer

State handling Services and Applications

 Statefullness in SIP Servers  Service creation possibilities with SIP
- Limitations of stateless servers - Overview of SIP-CGI, CPL, Java servlets, Jain,
- Transaction and dialog-stateful servers OSA/Parlay.
- Registration stateful servers. - Service examples with CPL
 The transaction layer  SIP Basic call-services and PBX-like features.
- Client and server transaction state machines - Call-forwarding, voicemail, CLIR/CLIP, etc
- The bugs and fixes for Non-INVITE transactions  Call-transfer and Call-Pickup
- REFER and Replaces:
Protocol extensions and updates  3rd party call control
 Extending the SIP protocol  SIP for events
- Using OPTIONS - The presence architecture
- Negotiation extensions - Triggering presence
- Requiring extensions - Dialog issues
- Handling new SIP methods in old proxies
 Instant messaging in SIP
 Reliable provisional responses
- Stand-alone messages with MESSAGE
- PRACK, RSeq and RAck - Session based messaging with MSRP
 Forking and Cancelling requests  B2BUA (Back to back User Agent)
- Cancel and stateful proxies - Requirements and Possibilities
 Caller preferences and Callee Capabilities
- Addressing and Registration extensions Exercise 4 – Services
 Using Early Media  Presence
- Simplex or Full-duplex  Instant messaging
- Issues with forking.
 Methods; SUBSCRIBE, NOTIFY, PUBLISH,
 Quality of Service and SIP/SDP MESSAGE

Exercise 3 – Forking and CANCEL Classic Telephony Using SIP

 Forking Calls – When and Why?  Sending DTMF in VoIP
 Response processing at forking. - DTMF and RTP, rfc2833/4733
 Method: CANCEL - INFO and SUBSRIBE/NOTIFY
 Parallel and sequential forking.  Merging ISUP/PSTN Networks and SIP
- Translating or tunneling ISUP
Security - SIP for telephones – SIP-T
- Q.1912.5– SIP-I
 Firewalling SIP servers and clients.
 E.164 Phone numbers and SIP-addresses
 Encryption and Authentication – How?
- Tel: and SIP: URIs
- Firewalls and NAT/PAT - Address translation, interworking.
- Session Traversal Utilities for NAT - DNS and ENUM.
- Traversal Using Relays around NAT
- SIP Outbound extension for NAT  Mobile SIP Telephony
- Symmetric Responses and Connection reuse. - IMS, IP Multimedia Subsystem
- Global Routable User-agent URI, GRUU
 SIP Privacy and Authenticity Exercise 5 – Security and DNS
- S/MIME examples for end to end security  User authentication/authorization and http digest
- Privacy and Anonymity
 Understanding DNS queries
 Securing the media channel
 Secure RTP with ZRTP or MIKEY
Links and references
 What to read first – List of RFC’s and Internet-Drafts.
 Links and references
 Acronym list

2 Spinnaker House, Juniper Drive, SW18 1FR London, U.K. T +46 768 633 899 E maziar.shoaie@gmail.com