Beskrivning
Detta är en avancerad SIP-kurs med mycket praktiska övningar. Funktioner såsom transaktionshantering, dialoger, diverse felsituationer, timers och mycket annat gås igenom i detalj. Du kommer att lära dig hur SIP fungerar inom både IP-telefoni och multimedia-lösningar, såsom presence och Instant Messaging (IM). Utbildningen går igenom hur SIP fungerar i mobila och fasta nät och grundkunskaper inom IP-telefoni och SIP behövs.
Målgrupp
Målgruppen för denna kurs är utvecklare och testare samt personer som jobbar med implementationer av SIP lösningar. Kursen är riktad både mot fasta och trådlösa lösningar.
Förkunskaper
Grundläggande kännedom om datakommunikation motsvarande kursen Datakommunikation Grundkurs, grundläggande kännedom om TCP/IP motsvarande kursen TCP/IP samt goda kunskaper i grundläggande VoIP och SIP motsvarande kursen SIP Fundamentals. Kunskapstest Testa dina kunskaper inom SIP gratis redan idag. Våra rådgivare kontaktar dig med förslag på vilken av våra utbildningar som är mest optimal för dig. SIP-test
Övrigt
Denna kurs finns som schemalagd utbildning och presentationen ges på svenska eller engelska. Under kursen blandas teoretiska presentationer med praktiska övningar. Vi kan även hålla denna kurs företagsintern. Kontakta oss för att få reda på hur vi kan hjälpa er med anpassade kurser.
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
- User authentication/authorization and http digest
- Understanding DNS queries
Links and references
- What to read first – List of RFC’s and Internet-Drafts.
- Links and references
- Acronym list
Utskriftsvänlig version
