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SIP Essentials
Code:
SIP
Duration:
5 Day
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$2595
USD
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Session Initiation Protocol (SIP) is the protocol uniting every communication management suite, be it Cisco Call Manager, Avaya Session and Communication Manager, Avaya IP Office, Oracle Session Border Controllers, Ericsson IMS cores, Asterisk, ShoreTel and Mitel products.
Youll make live call analyses with Wireshark and TCPDump. Via the PCAPs you create, as well as those accessed from an extensive library of premade captures, youll have no problems understanding why SIP makes the phone ring, how RTP carries real time voice and video, or troubleshooting and identifying errors.
The lessons in this course are clear and very technical. Attending students will receive access to the Alta3 Research SIP certification exam. Upon successful completion of the exam, students will be awarded a SIP certificate.
This course is available in the following formats:
Duration: 5 Day
Duration: 5 Day
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Delivery Format: Virtual Classroom
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$ 2595 |
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|
Delivery Format: Virtual Classroom
|
$ 2595 |
- 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
1. VoIP Introduction
- Circuit Switching
- VoIP Protocols
- VoIP Deployments: First Installations to Now
- SIP and the Softswitch
2. SIP Architecture
- The SIP Architecture
- UA, Proxy, Redirect, Forking, and B2BUA
- Multimedia Architecture
- RTP/RTCP
- SDP
- Methods
- REGISTER
- INVITE and ACK
- UPDATE
- OPTIONS
- REFER
- CANCEL
- SUBSCRIBE and NOTIFY
- MESSAGE
- BYE
- SIP Responses
- Via Path
- Record-Route
3. REGEX
- Regular Expression
4. Routing the SIP INVITE
- The Via: path
- Creation of Response-Path
- Response Merging
- Record-Route: and Route:
- Forking
- Loops and Spirals
5. The SIP Dialog
- The Purpose of the SIP Dialog
- How to Begin and End a Dialog
- The Dialog ID
6. SIP Entities
- B2BUA
- Proxy
- SBC
- Outbound Proxy
- UA
7. SIP Call Flow Examples
- The Following Call Flows Set Up and Examined Using Wireshark
- REGISTER
- Normal Call
- Busy
- Redirect
- Transfer (REFER)
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
9. SIP Uniform Resource Indicators (URIs)
- Generic URI information (RFC 2396)
- Direct or Proxy
- PSTN Number (RFC 2808)
- Instant Messaging
- Presence
- In Registrations
10. SIP Message Headers
- Via
- Branch
- Max-Forwards
- Dialog (To, From, and tag= fields, Call-ID)
- 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
11. Session Description Protocol (SDP)
- Session Parameters
- SDP Format
- Extending SDP
- SDPng
- Media Negotiation
- Changing Session Parameters
- Controlling the Media
12. SIP and the DNS
- Basic Resource Records (RR)
- A-Record, SOA, NS Record, MX 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)
13. ENUM
- ENUM Protocol (RFC 3761)
- Dynamic Delegation Discovery System (RFC 3401, 3402, 3403, 3761, 3764)
- How SIP Uses ENUM
14. SIP and DHCP
- DHCP Protocol
- SIP DHCP Options
15. Interoperating SIP with Legacy PSTN Signaling
- Call Transfer (REFER)
- 183 Early Media
- Interworking SIP with Local Call Control (E&M or DID)
- SIP and the PSTN
- SIP-T
16. RTP and Real-Time Control Protocol (RTCP)
- Dealing 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
17. DTMF Handling
- Inband
- RFC 2833
- SIP INFO
18. Fax Handling
- Inband
- Fax Relay
- T.38
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
20. SIP Timers
- T1, T2, T4
- Timer A K
21. SIP Security
- Security for Call Setup
- Authentication
- S/MIME
- TLS
22. NAT Traversal
- How NAT Operates on SIP and SDP
- NAT Types
- STUN
- TURN
- ICE
23. SIPp: A SIP Testing Tool
- SIPp
- SIPp XML Examples
1. VoIP Introduction
- Circuit Switching
- VoIP Protocols
- VoIP Deployments: First Installations to Now
- SIP and the Softswitch
2. SIP Architecture
- The SIP Architecture
- UA, Proxy, Redirect, Forking, and B2BUA
- Multimedia Architecture
- RTP/RTCP
- SDP
- Methods
- REGISTER
- INVITE and ACK
- UPDATE
- OPTIONS
- REFER
- CANCEL
- SUBSCRIBE and NOTIFY
- MESSAGE
- BYE
- SIP Responses
- Via Path
- Record-Route
3. REGEX
- Regular Expression
4. Routing the SIP INVITE
- The Via: path
- Creation of Response-Path
- Response Merging
- Record-Route: and Route:
- Forking
- Loops and Spirals
5. The SIP Dialog
- The Purpose of the SIP Dialog
- How to Begin and End a Dialog
- The Dialog ID
6. SIP Entities
- B2BUA
- Proxy
- SBC
- Outbound Proxy
- UA
7. SIP Call Flow Examples
- The Following Call Flows Set Up and Examined Using Wireshark
- REGISTER
- Normal Call
- Busy
- Redirect
- Transfer (REFER)
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
9. SIP Uniform Resource Indicators (URIs)
- Generic URI information (RFC 2396)
- Direct or Proxy
- PSTN Number (RFC 2808)
- Instant Messaging
- Presence
- In Registrations
10. SIP Message Headers
- Via
- Branch
- Max-Forwards
- Dialog (To, From, and tag= fields, Call-ID)
- 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
11. Session Description Protocol (SDP)
- Session Parameters
- SDP Format
- Extending SDP
- SDPng
- Media Negotiation
- Changing Session Parameters
- Controlling the Media
12. SIP and the DNS
- Basic Resource Records (RR)
- A-Record, SOA, NS Record, MX 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)
13. ENUM
- ENUM Protocol (RFC 3761)
- Dynamic Delegation Discovery System (RFC 3401, 3402, 3403, 3761, 3764)
- How SIP Uses ENUM
14. SIP and DHCP
- DHCP Protocol
- SIP DHCP Options
15. Interoperating SIP with Legacy PSTN Signaling
- Call Transfer (REFER)
- 183 Early Media
- Interworking SIP with Local Call Control (E&M or DID)
- SIP and the PSTN
- SIP-T
16. RTP and Real-Time Control Protocol (RTCP)
- Dealing 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
17. DTMF Handling
- Inband
- RFC 2833
- SIP INFO
18. Fax Handling
- Inband
- Fax Relay
- T.38
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
20. SIP Timers
- T1, T2, T4
- Timer A K
21. SIP Security
- Security for Call Setup
- Authentication
- S/MIME
- TLS
22. NAT Traversal
- How NAT Operates on SIP and SDP
- NAT Types
- STUN
- TURN
- ICE
23. SIPp: A SIP Testing Tool
- SIPp
- SIPp XML Examples
Lab 1: Construct and Enable a VoIP Network
Lab 2: SIP User Agent Configuration
Lab 3: Direct UA to UA Routing with No Proxy
Lab 4: Proxy Based SIP Routing
Lab 5: Adding Authorized UAs to a Domain
Lab 6: Intra Domain Routing (SIP in the Same Domain)
Lab 7: SIP REGISTER Registering a SIP UA
Lab 8: Registering a SIP UA Soft Client
Lab 9: Registering a SIP UA Client to a Mobile Device
Lab 10: Inter Domain Routing (SIP in Different Domains)
Lab 11: Strip off the Leading 9
Lab 12: PDT Management
Lab 13: Using Wireshark
Lab 14: Capture a SIP Registration via Wireshark
Lab 15: Capture a Normal SIP Call via Wireshark
Lab 16: Capture a Call to a Vacant Number via Wireshark
Lab 17: Capture a SIP Call to Busy Number via Wireshark
Lab 18: Capture a Call Forward via Wireshark
Lab 19: Via, Record Route, and Route Headers
Lab 20: Examining Max Forwards
Lab 21: INVITE with SDP sendonly vs. sendrecv
Lab 22: Silence Suppression
Lab 23: DTMF RFC 2833 and SIP INFO
Lab 24: SIP B2BUA Configuration Example
Lab 25: Register Linksys SIP Phone with Asterisk PBX
Lab 26: SIP Presence (NOTIFY)
Lab 27: RTP Relay
Lab 28: Direct RTP Flow between Two UAs 3PCC
Lab 29: ENUM Call Routing
Lab 30: Testing SIP Connectivity Using SIP OPTIONS
Lab 31: Advanced: SIP Testing with SIP-p
Lab 1: Construct and Enable a VoIP Network
Lab 2: SIP User Agent Configuration
Lab 3: Direct UA to UA Routing with No Proxy
Lab 4: Proxy Based SIP Routing
Lab 5: Adding Authorized UAs to a Domain
Lab 6: Intra Domain Routing (SIP in the Same Domain)
Lab 7: SIP REGISTER Registering a SIP UA
Lab 8: Registering a SIP UA Soft Client
Lab 9: Registering a SIP UA Client to a Mobile Device
Lab 10: Inter Domain Routing (SIP in Different Domains)
Lab 11: Strip off the Leading 9
Lab 12: PDT Management
Lab 13: Using Wireshark
Lab 14: Capture a SIP Registration via Wireshark
Lab 15: Capture a Normal SIP Call via Wireshark
Lab 16: Capture a Call to a Vacant Number via Wireshark
Lab 17: Capture a SIP Call to Busy Number via Wireshark
Lab 18: Capture a Call Forward via Wireshark
Lab 19: Via, Record Route, and Route Headers
Lab 20: Examining Max Forwards
Lab 21: INVITE with SDP sendonly vs. sendrecv
Lab 22: Silence Suppression
Lab 23: DTMF RFC 2833 and SIP INFO
Lab 24: SIP B2BUA Configuration Example
Lab 25: Register Linksys SIP Phone with Asterisk PBX
Lab 26: SIP Presence (NOTIFY)
Lab 27: RTP Relay
Lab 28: Direct RTP Flow between Two UAs 3PCC
Lab 29: ENUM Call Routing
Lab 30: Testing SIP Connectivity Using SIP OPTIONS
Lab 31: Advanced: SIP Testing with SIP-p
Any company or individual who wants to advance their comprehension of VoIP and SIP