Combined SIP Advanced and SIP Troubleshooting

Agenda

Day 1-3

VoIP Signalling Overview

• Functional components of a Voice over IP Network
• The need for VoIP signalling and the different alternatives
  • SIP, H.323, MEGACO, BICC
  • IETF Multimedia Architecture
  • Signalling and User plane separation
• Brief repetition of RTP and RTCP

SIP Refresher and SDP Update

• 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
• Session Description Protocol
  • SDP Offer/Answer Model
  • Quality of Service Extensions
  • Connection Oriented Transports in SDP
  • Media groupings in SDP

Basic Calls

• Basic SIP sessions with SIP proxy and Registrar
• Using Wireshark for traffic monitoring
• SIP Signalling and SDP Negotiation analysis
• Methods; INVITE, BYE, REGISTER

Protocol Foundation

• Message structure and format rules
  • Mandatory headers and parameters.
• Proxy and Client DNS Usage
  • NAPTR record type
  • SRV record type
  • A record type
• SIP message routing rules
  • Route headers
  • Record Routing
  • Via header response routing
• Detailed proxy behaviour
  • Location server lookup
  • Request forwarding
  • Response processing
• Statefullness in SIP Servers
  • Limitations of stateless servers
  • Transaction and dialog-stateful servers
  • Registration stateful servers
• Creating early dialogs for early media

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

Step by step walkthrough of advanced call scenario Features and Functionality

• 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
  • Require Qos with SDP parameters
  • Using UPDATE in early dialogs

Forking and CANCEL

• Forking Calls When and Why?
• Response processing at forking
• Method: CANCEL
• Parallell and sequential forking

Security

• Firewalling SIP servers and clients.
• Encryption and Authentication – How?
• The Firewall and NAT problem
  • SIP-away firewalls
  • SIP signalling and NAT
  • Symmetric Responses
  • Managing Client Initiated Connections in SIP
  • Media NAT traversal: STUN, TURN, ICE
• SIP Privacy and Authenticity
  • S/MIME examples for end to end security
  • Privacy services
• Securing the media channel

Security and DNS

• User authentication and http digest
• SIP symmetric responses
• Understanding DNS queries

Services and Applications

• Service creation possibilities with SIP
  • Overview of SIP-CGI, CPL, Java servlets, Jain, 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
• SIP and presence
• 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
• Authorization and message encoding
• Methods; SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE

Classic Telephony Using SIP

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

Summary and Future

• Summarizing the whole course
• What to read first – List of RFC’s and Internet-Drafts
• Links and references

Day 4-5

Troubleshooting Basics

• Troubleshooting techniques
  -        Generating baselines
  -        Documenting the system
  -        Reference Models
  -        Isolating the error

Troubleshooting Tools

• Overview of useful troubleshooting tools
  -        Wireshark, Tcpdump
  -        SipSAK, SIPp, SIP Scenario Generator
  -        Ntop, MRTG,
  -        DNS, dig and nslookup.

• Practical Exercise 1

SIP Troubleshooting

• Finding the rules
  -        IETF tools, IETF Workgroups
  -        The simple SIP Checklist

• Important SIP Rules
  -        Invite vs Non Invite
  -        SDP Usage
  -        Statehandling
  -        Message Parsing
  -        Registration and Caller preferences

Detailed Troubleshooting

Dialog Management

• Rules for Dialog state management
  -        Creating, modifying, releasing
  -        Recording Routes and Route-set management
  -        Early dialogs, early media, and PRACK.

• Practical Exercise 3

Message Forwarding

• More routing and Record-Route
• Loops, Forking, and CANCEL
• Transactions and the statemachines
  -        Updates to the state machines, rfc4320/21
  -        SIP Timer

• Practical Exercise 4

DNS and Transport

• The rules for selecting transport protocol
  -        Message sizes, URI analysis, and DNS

Summary

• 
  Taisto Qvist