A3SIP: SIP Essentials

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About this Course

Session Initiation Protocol (SIP) is the protocol uniting every communication management suite, be it Cisco Call Manager, Avaya Session and Communication Manager, Avaya IP Office, Oracle Session Border Controllers, Ericsson IMS cores, Asterisk, ShoreTel and Mitel products. You’ll make live call analyses with Wireshark and TCPDump. Via the PCAPs you create, as well as those accessed from an extensive library of premade captures, you’ll have no problems understanding why SIP makes the phone ring, how RTP carries real time voice and video, or troubleshooting and identifying errors.
The lessons in this course are clear and very technical. Attending students will receive access to the Alta3 Research SIP certification exam. Upon successful completion of the exam, students will be awarded a SIP certificate.

Full outline here: https://alta3.com/courses/sip

Audience Profile

Any company or individual who wants to advance their comprehension of VoIP and SIP

At Course Completion

  • SIP Requests and Responses
  • Live call capture
  • Wireshark Analysis (pcaps & ng-pcap)
  • RTP Voice and Video
  • Session Description Protocol (SDP) negotiation
  • DTMF transmission
  • SIP Routing and Dialplan construction (regular expression)
  • Call flow analysis
  • Testing with SIP-p
  • Troubleshooting (failed calls, 1-way or no way voice)
  • STUN / TURN / ICE

Outline

AI LLM Toolkit
• 💻 Lecture + Lab: Large Language Model toolkit for AI Solution Assistance
VOIP Fundamentals
• 💬 Lecture: Introduction to VoIP
Packet Captures
• 💻 Lecture + Lab: Upload PCAPs for analysis (OPTIONAL)
• 💻 Lecture + Lab: Introduction to Wireshark
• 💻 Lecture + Lab: Termshark
SIP Registrars
• 💬 Lecture: SIP Architecture
• 💻 Lecture + Lab: Successful REGISTER by a User Agent
• 💻 Lecture + Lab: REGISTER Fails Auth
• 💻 Lecture + Lab: deREGISTER Log Out
SIP INVITE
• 💬 Lecture: Regular Expression
• 💻 Lecture + Lab: Building a Dial Plan with RegEx
• 💬 Lecture: Routing the INVITE
• 💻 Lecture + Lab: The SIP INVITE
• 💻 Lecture + Lab: SIP INVITE Packet Analysis with Wireshark
Establishing Calls
• 💬 Lecture: SIP Dialog
• 💻 Lecture + Lab: Troubleshooting Common SIP Failures with Wireshark
• 💬 Lecture: SIP Entities
Call Flows
• 💬 Lecture: Basic SIP Call Flows
• 💬 Lecture: SIP 3xx Redirection
• 💻 Lecture + Lab: Call Forwarding or 3xx responses
• 💬 Lecture: SIP REFER
• 💻 Lecture + Lab: SIP REFER for Call Transfer
• 💬 Lecture: SIP PRACK 100rel
• 💻 Lecture + Lab: SIP PRACK 100rel
• 💬 Lecture: Call Forking
• 💻 Lecture + Lab: Call Forking
SIP Proxies
• 💬 Lecture: Call Routing
• 💻 Lecture + Lab: INVITE Relay by SIP Proxies
• 💻 Lecture + Lab: No Record Routes
• 💬 Lecture: SIP URIs
• 💻 Lecture + Lab: CANCELed SIP call
• 💻 Lecture + Lab: Global Failures or 6xx responses
Supporting Systems
• 💬 Lecture: SIP and the DNS
• 💬 Lecture: ENUM
• 💬 Lecture: Interop with the PSTN
SIP Tools
• 💻 Lecture + Lab: Install Asterisk
• 💻 Lecture + Lab: Linux Fundamentals
• 💻 Lecture + Lab: Using vim
• 💻 Lecture + Lab: Making pcaps with tcpdump
• 💻 Lecture + Lab: Making pcaps with tshark
• 💬 Lecture: SIPp
• 💻 Lecture + Lab: SIPp SIP Tester
• 💻 Lecture + Lab: SIP Swiss Army Knife
SIP Headers
• 💬 Lecture: Common SIP Headers
Session Description Protocol
• 💬 Lecture: Session Description Protocol
• 💻 Lecture + Lab: Session Description Protocol
• 💻 Lecture + Lab: SDP Video Call Setup
• 💻 Lecture + Lab: SDP Video Call Setup Fails
Real-Time Transport Protocol
• 💬 Lecture: Real-time Transport Protocol
• 💻 Lecture + Lab: One-Way Media
Dual Tone Multi Frequency
• 💬 Lecture: Transmitting DTMF
• 💻 Lecture + Lab: Methods for Transport of DTMF
Fax
• 💬 Lecture: Fax Handling
Presence
• 💬 Lecture: Presence
• 💻 Lecture + Lab: SIP PUBLISH
• 💻 Lecture + Lab: Presence and IM Exchange
SIP Timers
• 💬 Lecture: SIP Timers
SIP Security
• 💬 Lecture: SIP Security
NAT Issues
• 💬 Lecture: NAT

Prerequisites

None