Course description
SIP Essentials
In this SIP course, students learn Session Initiation Protocol and important protocols related to SIP implementations. This course thoroughly explains what SIP is, how it works, and also provides a practical guide on how to use it. The lessons in this course are clear and very technical. In this course, students will examine how SIP interoperates in the current telecommunications network, going beyond the basics of the protocol and getting a big picture understanding of how it all fits together. At the end of the course, students will receive access to an Alta3 Research SIP certification exam. Upon successful completion of the exam, students will be awarded a SIP certificate.
Upcoming start dates
Training content
Day 1 - SIP Architecture
Ch 1. VoIP Introduction
- Circuit Switching
- VoIP Protocols Overview
- VoIP Deployments from the First Installations to Now
- SIP and the Softswitch
Ch 2. SIP Architecture
- The SIP Architecture
- UA, Proxy, Redirect, Forking, B2BUA
- Multimedia Architecture
- RTP/RTCP
- SDP
- Methods: REGISTER, INVITE and ACK, UPDATE OPTIONS, CANCEL, REFER, SUBSCRIBE and NOTIFY, MESSAGE, BYE
- SIP Responses
- Via Path
- Record-route
Day 2 - Understanding the SIP Dialog
Ch 3. REGEX
- Regular Expression
- Building SIP Dialplans with REGEX
Ch 4. Routing the SIP INVITE
- The Via: path
- Creation of Response-Path
- Response Merging
- Record-route and Route:
- Forking
- Loops and Spirals
Ch 5. The SIP Dialog
- The Purpose of the SIP Dialog
- How to Begin and End a Dialog
- The Dialog ID
Ch 6. SIP Entities
- User Agents
- Back-to-Back UAs
- Proxy
- Session Border Controller
- Outbound Proxies
Day 3 - Advanced SIP Messaging
Ch 7. SIP Call Flows Examples
- REGISTER
- Normal call
- Busy
- Redirect
- Transfer (REFER)
Ch 8. SIP Call Routing
- How SIP Routing is Used to Route CALLS
- Use of Record-Route in Stateless Routing Proxies
- How SIP is Used in the PSTN Migration to An All IP Network
Ch 9. SIP Uniform Resource Indicators (URIs)
- Generic URI Information (RFC 3986)
- Direct or Proxy
- PSTN Number (RFC 2808)
- Instant Messaging
- Presence
- In Registrations
Ch 10. SIP Message Headers
- SIP Dialog (To:, From:, tag= fields, Call-ID:)
- Via: & Branch
- Max-Forwards:
- CSeq:
- Proxy-Authenticate:
- Proxy-Authorize:
- Contact:
- Expires:
- User-Agent:
- Content-Length:
- Allow: Supported:
- P-Access-Network-Info
- P-Charging-Vector:
- P-Preferred-Identity:
- P-Asserted-Identity:
- Authorization:
- Security-Client:
- Security-Server:
- Content-Type:
Day 4 - Session Description Protocol, Real-time Transport Protocol, and Legacy Interop
Ch 11. Session Description Protocol (SDP)
- Session Parameters
- SDP Format
- Extending SDP
- SDPng
- Media Negotiation
- Changing Session Parameters
- Controlling the Media
Ch 12. SIP and the DNS
- Basic Resource Records (RR)
- A-record, SOA, NS Record, MX Record (Important for Comparison to the SRV Record)
- The SRV Record (RFC 2782)
- How SIP Uses the SRV Record (RFC 3263 Locating SIP servers)
- How to Configure a SRV Record
- The NAPTR Record (RFC 2915)
Ch 13. ENUM
- ENUM Protocol RFC 3761
- Dynamic Delegation Discovery System (RFC 3401, 3402, 3403, 3761, 3764)
- How SIP Uses ENUM
Ch 14. SIP and DHCP
- DHCP Protocol
- SIP DHCP Options
Ch 15. Interoperating SIP with Legacy STN Signaling
- Call Transfer (REFER)
- 183 Early Media
- Interworking SIP with Local Call Control (E&M or DID)
- SIP and the PSTN SIP-T
Ch 16. Real-time Transport Protocol (RTP) and Real-time Control Protocol (RTCP)
- Dealing with Packet Loss, Latency & Jitter
- How RTP Defines the Session
- Session Description Protocol
- The RTP Profile
- The RTP Payload Type Field
- RTP Telephony Events (RFC 2833)
- How RTP Removes Jitter
- How RTP Handles Packet Loss
- How RTP Identifies the Talking Party
- How RTP Handles Silence Suppression
- How RTP Handles Fixed Length Packets (Padding)
- How RTP is Used to Mix Voice (Conference Calls)
- The RTP Header
- RFC 2833 Protocol
- RTP Control Protocol
- SDES
- Sender/Receiver Reports
- Bye Reports
Day 5 - Applications of SIP and Troubleshooting
Ch 17. DTMF Handling
- Inband
- RFC 2833
- SIP INFO
Ch 18. Fax Handling
- Inband
- Fax Relay
- T.38
Ch 19. Presence
- SIMPLE: SIP for Instant Messaging and Presence Leveraging Extensions
- Terminology
- Framework
- Resource List Manipulation Requirements
- Authorization Policy Manipulation
- Acceptance Policy Requirements
- Notification Requirements
- Content Requirements
- General Requirements
Ch 20. SIP Timers
- T1, T2, T4
- Timer A-K
Ch 21. SIP Security
- Security for Call Setup
- Authentication
- S/MIME
- TLS
Ch 22. SIP NAT Traversal
- How NAT operates on SIP and SDP
- NAT Types
- STUN
- TURN
- ICE
Ch 23. SIPp: A SIP Testing Tool
- SIPp
- SIPp XML Examples
Costs
- Live Webinar fee: $2,495.00
- E-Learning fee: $295.00
- Contact the provider for more information about pricing for on-site delivery option
Certification / Credits
At the end of the course, students will receive access to an Alta3 Research SIP certification exam. Upon successful completion of the exam, students will be awarded a SIP certificate.
Why choose Alta3?
25+ years specializing in technology training
Over 95% student satisfaction rate
Training delivered in 9+ countries
Reviews
Average rating 5
Thank you for your patience and your expert knowledge in explaining everything. I will definitely need to go back to the slide decks and videos for reviews as this is fairly ne...
Zach went above and beyond my expectations of an instructor. He demonstrated a concrete understanding of the material and was able to pass that knowledge back to us in an entert...
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Alta3 Research
Since 1997, Alta3 Research has been empowering organizations and individuals using their expert approach to Cloud, DevOps and Telecom training. We develop and perform technical training via live classroom, online classes and our library of self-paced learning videos and labs....
I had some reservations about the virtual learning environment. I am glad to say I found this experience, possibly because of Zach's pace, and teaching style actually more desir...