SIP Advanced
This is an advanced SIP training with practical exercises. Functions such as transaction handling, dialogs, different error situations, timers and a lot of other topics are covered in detail.
eLearning
€ 595*
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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.
This is a theoretical training with practical exercises.
Who should attend?
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.
Course outline
- 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
- Message structure and format rules.
- Mandatory headers and parameters.
- Addressing and using URIs
- SIP Registration
- Expiration and Deregistration
- The location service
- SIP message routing rules
- Request-URI and Route headers
- Via header response routing
- Route and Record-Route
- Detailed proxy behavior
- Location server lookup
- Request forwarding
- Response processing
- Loop detection/prevention
- Using DNS for inter-domain signaling
- SIP related DNS records
- Load sharing and redundancy features
- Establishing sessions with INVITE
- Using SDP for negotiation
- SDP Offer/Answer Model
- Dialog creation
- Early Dialogs and UPDATE
- Dialog state management
- Statefullness in SIP Servers
- Transaction and dialog-stateful servers
- The transaction layer
- Client and server transaction state machines
- Extending the SIP protocol
- Negotiation extensions
- Reliable provisional responses
- Forking and Cancelling requests
- Using Early Media
- Quality of Service and SIP/SDP
- Firewalling SIP servers and clients.
- Encryption and Authentication
- Firewalls and NAT/PAT
- Session Traversal Utilities for NAT
- Traversal Using Relays around NAT
- SIP Privacy and Authenticity
- SIP Basic call-services and PBX-like features.
- Call-transfer and Call-Pickup
- REFER and Replaces
- 3rd party call control
- SIP for events
- Instant messaging in SIP
- B2BUA (Back to back User Agent)
- Sending DTMF in VoIP
- Merging ISUP/PSTN Networks and SIP
- 164 Phone numbers and SIP-addresses
- DNS and ENUM
- IMS, IP Multimedia Subsystem
- Standard phone call
- Important headers
- Dialogs
- Transaction matching
- Registration issues
- Inter-domain call setup and routing
- DNS usage
- Record-routing
- Dialog creation
- Choosing codec
- SDP details
- ACK handling variations
- Forking Calls
- Response processing at forking
- CANCEL
- Identifying extensions
- Determining support for SIP functions
- PRACK usage
- QoS negotiations
- Digest authentication
- Presence
- Instant messaging
- Initial vs Re-INVITE
- Service sessions and dialogs