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Overview |
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In this hands on workshop you will develop an expert knowledge of Session Initiation Protocol. Learn the latest, up to the minute developments with this suite of critically important Internet Engineering Task Force standards. Join your colleagues for an intense, hands on workshop covering the latest SIP developments. Through Real World examples and Hands ON labs you develop deep VoIP/SIP domain experience that is perfectly suited to your next design project. |
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VoIP QoS Design Fundamentals |
- VoIP Quality Overview
- Bandwidth & Real Time Requirements Defined
- Network & Considerations:
- Delay
- Jitter
- Packet Loss
- Echo & Echo Cancellation
- Codec Selection: Impact on QoS
- Quality Of Service Protocols
- QOS related networking protocols
- Integrated Serives & Resource Reservation Protocol (RSVP)
- Differentiated Services (DiffServ)
- Multiprotocol Label Switching (MPLS
- QoS Design with Cisco specific WAN/LAN Infrastructure
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The Session Initiation Protocol: SIP Architecture |
- History and development
- H.323 v/s SIP
- SIP overview
- SIP components
- SIP pros and cons
- Standards for SIP and related protocols
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SIP: Protocol Operation |
- H.323 v/s SIP
- SIP overview
- SIP components
- SIP Pros & Cons
- Standards for SIP and related protocols
- IETF Development
- Current State of IETF Standards
- IETF Standards trends and Design Considerations
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SIP Protocol & Operation |
- Client/Server Transactions
- Proxy Servers
- SIP Messages
- Transport Layer
- Examples
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SIP Programming |
- SIP Programming
- LAB(s): SIP programming using C/C++ library
- Use open source C++ SIP library reSIProcate to create SIP client
- Understand the FSM for INVITE
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- SIP programming using C/C++ library
- Use open source C++ SIP library reSIProcate to create SIP client
- Use FSM for INVITE
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Extending SIP |
- Overview
- Extension Negotiation
- Technical details of SIP extensions
- SIP Extensions
- SDP details
- SS7 / SIP Signaling Integration Issues
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Advanced Topics in SIP |
- Security
- Loose Routing in RFC 3261
- Firewalls & SIP
- Testing
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SIP Applications |
- SIP call flows
- Peer-to-peer calls
- UA to UA through the same SIP proxy server
- UA to UA from one domain to another domain through SIP proxy servers
- SUBSCRIBE & NOTIFY Call Flow for Instant Messaging
- Blind Call transfer
- Consultative call transfer
- Third party call control
- SIP to PSTN Call flow
- PSTN to SIP Call Flow
- SIP to H.323 Call Flow
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SIP Testing |
- SIP testing overview
- SIP Testing overview
- Interoperability Testing using Asterisk
- LAB: PROTOS test suite for SIP functionality
- SIP testing standards
- ETSI
- SIPp
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