SIP Essentials

SIP EssentialsSIPA3Alta3A3-SIP1None RequiredAny company or individual who wants to advance their comprehension of VoIP and SIP.1. SIP Introduction <ul> <li>SIP Message Format</li><li>Legacy Call Control</li><li>Compare SIP</li><li>Packetizing Voice</li><li>SIP Call Flow</li><li>How SIP Routes Media</li><li>SIP Call Control</li><li>SIP in 4G</li></ul>2. SIP Architecture <ul> <li>SIP UA</li><li>SIP Requests</li><li>SIP Response</li><li>SIP URI</li><li>SIP Architecture</li><li>SIP Domain</li><li>SIP Registration</li><li>SIP Call Routing</li><li>Loose Routing</li></ul>3. Regular Expression <ul> <li>Metacharacters</li><li>Substitution</li><li>REGEX Modifications</li></ul>4. Routing the SIP INVITE <ul> <li>Proxy Routing</li><li>Via and Record-Route</li></ul>5. The SIP Dialog <ul> <li>SIP Dialog</li><li>The reINVITE</li></ul>6. SIP Entities <ul> <li>SIP Topology</li><li>SIP Proxy</li><li>B2BUA</li><li>Outbound Proxy</li></ul>7. SIP Call Flow Examples <ul> <li>Wireshark Colors</li><li>Wireshark Preferences</li><li>SIP Stack</li><li>REGISTER with Authentication</li><li>Wireshark Analysis of SIP Dialog</li><li>SIP Redirect</li><li>CFNA</li><li>REFER and Call Transfer</li></ul>8. SIP Call Routing <ul> <li>PRACK 100-rel</li><li>Call Forking</li><li>Loop and Spiral</li><li>Third Party Call Control</li><li>Path Minimalization</li><li>SIP in the PLMN</li><li>OPTIONS Method</li></ul>9. SIP Uniform Resource Indicators (URIs) <ul> <li>URI vs. URL vs. URN</li><li>SIP URI Examples</li><li>URI Delimeters</li><li>SIP and SIPs</li><li>tel URI</li><li>URI Escape Codes</li></ul>10. SIP Message Headers <ul> <li>SIP Header Overview</li><li>Dialog ID Headers</li><li>User-Agent</li><li>SIPp Header Modification</li><li>Proxy-Authenticate</li><li>Allow and Supported</li><li>History Info</li><li>Join</li><li>Session Expires</li><li>PPI and PIA</li><li>Establish Service Path</li><li>IMS Hosted</li><li>Content-Type</li></ul>11. Session Description Protocol (SDP) <ul> <li>SDP Background</li><li>SDP Format</li><li>SIP = one way?</li><li>SDP Lines</li><li>SDP Offer/Answer</li><li>Call Hold</li></ul>12. SIP and the DNS <ul> <li>Zone File</li><li>SOA and NS Records</li><li>A-Record</li><li>SRV Record</li><li>NAPTR Record</li><li>Locating SIP Servers</li></ul>13. ENUM <ul> <li>ENUM Database Example</li><li>ENUM Query and Response</li><li>ENUM REGEX</li><li>Post ENUM Routing</li></ul>14. Legacy <ul> <li>Early Media</li><li>SIP-T and SIP-I</li></ul>15. RTP and Real-Time Control Protocol (RTCP) <ul> <li>RTP Headers</li><li>RTP Dejitter</li><li>Conferencing</li><li>RTCP</li></ul>16. DTMF Handling <ul> <li>DTMF</li><li>SIP INFO</li><li>RFC 2833</li></ul>17. Fax Handling <ul> <li>T.30</li><li>T.38</li><li>SDP RFC 3407</li></ul>18. Presence <ul> <li>Presence Overview</li><li>PIDF XML Example</li><li>Rich Presence</li><li>Presence Message Flow</li><li>Instant Messaging</li></ul>19. SIP Timers <ul> <li>Standard Timer Values</li><li>Session-Expires</li></ul>20. SIP Security <ul> <li>Security for Call Setup</li><li>Authentication</li><li>S/MIME</li><li>TLS</li></ul>21. SIP NAT Traversal <ul> <li>NAT</li><li>NAT Types</li><li>STUN & TURN</li></ul>SIP Essentials Labs <ul> <li>Understanding the Network</li><li>Lab 1: Construct & Enable a VoIP Network</li><li>Lab 2: SIP User Agent Configuration</li><li>Lab 3: Direct UA to UA Routing with No Proxy</li><li>Lab 4: Proxy Based SIP Routing</li><li>Lab 5: Adding Authorized UAs to a Domain</li><li>Lab 6: Intra Domain Routing (SIP in the same domain)</li><li>Lab 7: SIP REGISTER – Registering a SIP UA</li><li>Lab 8: Registering a SIP UA Softclient</li><li>Lab 9: Registering a SIP UA Client to a Mobile Device</li><li>Lab 10: Inter Domain Routing (SIP in different domains)</li><li>Lab 11: Strip off the Leading ‘9’</li><li>Lab 12: PDT Management</li><li>Lab 13: Using Wireshark</li><li>Lab 14: Capture a SIP Registration via Wireshark</li><li>Lab 15: Capture a ‘Normal’ SIP Call via Wireshark</li><li>Lab 16: Capture a Call to a Vacant Number via Wireshark</li><li>Lab 17: Capture a SIP Call to Busy Number via Wireshark</li><li>Lab 18: Capture a Call Forward via Wireshark</li><li>Lab 19: Via, Record Route, and Route Headers</li><li>Lab 20: Examining ‘Max Forwards’</li><li>Lab 21: INVITE with SDP – sendonly vs. sendrecv</li><li>Lab 22: Silence Suppression</li><li>Lab 23: DTMF RFC 2833 and SIP INFO</li><li>Lab 24: SIP B2BUA Configuration Example</li><li>Lab 25: Register Linksys SIP Phone with Asterisk PBX</li><li>Lab 26: SIP Presence (NOTIFY)</li><li>Lab 27: RTP Relay</li><li>Lab 28: Direct RTP Flow Between Two UAs – 3PCC</li></ul>None RequiredAny company or individual who wants to advance their comprehension of VoIP and SIP.1. SIP Introduction - SIP Message Format - Legacy Call Control - Compare SIP - Packetizing Voice - SIP Call Flow - How SIP Routes Media - SIP Call Control - SIP in 4G 2. SIP Architecture - SIP UA - SIP Requests - SIP Response - SIP URI - SIP Architecture - SIP Domain - SIP Registration - SIP Call Routing - Loose Routing 3. Regular Expression - Metacharacters - Substitution - REGEX Modifications 4. Routing the SIP INVITE - Proxy Routing - Via and Record-Route 5. The SIP Dialog - SIP Dialog - The reINVITE 6. SIP Entities - SIP Topology - SIP Proxy - B2BUA - Outbound Proxy 7. SIP Call Flow Examples - Wireshark Colors - Wireshark Preferences - SIP Stack - REGISTER with Authentication - Wireshark Analysis of SIP Dialog - SIP Redirect - CFNA - REFER and Call Transfer 8. SIP Call Routing - PRACK 100-rel - Call Forking - Loop and Spiral - Third Party Call Control - Path Minimalization - SIP in the PLMN - OPTIONS Method 9. SIP Uniform Resource Indicators (URIs) - URI vs. URL vs. URN - SIP URI Examples - URI Delimeters - SIP and SIPs - tel URI - URI Escape Codes 10. SIP Message Headers - SIP Header Overview - Dialog ID Headers - User-Agent - SIPp Header Modification - Proxy-Authenticate - Allow and Supported - History Info - Join - Session Expires - PPI and PIA - Establish Service Path - IMS Hosted - Content-Type 11. Session Description Protocol (SDP) - SDP Background - SDP Format - SIP = one way? - SDP Lines - SDP Offer/Answer - Call Hold 12. SIP and the DNS - Zone File - SOA and NS Records - A-Record - SRV Record - NAPTR Record - Locating SIP Servers 13. ENUM - ENUM Database Example - ENUM Query and Response - ENUM REGEX - Post ENUM Routing 14. Legacy - Early Media - SIP-T and SIP-I 15. RTP and Real-Time Control Protocol (RTCP) - RTP Headers - RTP Dejitter - Conferencing - RTCP 16. DTMF Handling - DTMF - SIP INFO - RFC 2833 17. Fax Handling - T.30 - T.38 - SDP RFC 3407 18. Presence - Presence Overview - PIDF XML Example - Rich Presence - Presence Message Flow - Instant Messaging 19. SIP Timers - Standard Timer Values - Session-Expires 20. SIP Security - Security for Call Setup - Authentication - S/MIME - TLS 21. SIP NAT Traversal - NAT - NAT Types - STUN & TURN SIP Essentials Labs - Understanding the Network - Lab 1: Construct & 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 Softclient - 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 – 3PCC5 jours2795.003855.00