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SIP

SIP. South Carolina Cisco User Group. Martin Jefferson IE UC Practice Manager. Agenda. What is SIP? Why SIP over T1/PRI? SIP Trunking Cisco SIP End points. Agenda. What is SIP? Why SIP over T1/PRI? SIP Trunking Cisco SIP End points. What is SIP?.

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SIP

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  1. SIP South Carolina Cisco User Group Martin Jefferson IE UC Practice Manager

  2. Agenda • What is SIP? • Why SIP over T1/PRI? • SIP Trunking • Cisco SIP End points

  3. Agenda • What is SIP? • Why SIP over T1/PRI? • SIP Trunking • Cisco SIP End points

  4. What is SIP? • Session Initiation Protocol (RFC 3261) • SIP uses Text based commands • SIP User Agent (UA) • SIP User Agent Client (UAC) & User Agent Server (UAS) • Session Border Controllers

  5. SIP • The Session Initiation Protocol (SIP) is an IETF-defined signaling protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP). • SIP clients typically use TCP or UDP on port numbers 5060 and/or 5061 to connect to SIP servers and other SIP endpoints

  6. SIP

  7. SIP Messages • Invite INVITE sip:5620@172.22.5.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.22.12.3:5060;branch=z9hG4bK155691746 Remote-Party-ID: "JEFFERSON MARTI" <sip:8432762180@172.22.12.3>;party=calling;screen=yes;privacy=off From: "JEFFERSON MARTI" <sip:8432762180@172.22.12.3>;tag=B064C8D4-DF3 To: <sip:5620@172.22.5.10> Date: Wed, 28 Jul 2010 15:38:36 GMT Call-ID: 54322D8-999511DF-99D69AFE-2081E384@172.22.12.3 Supported: timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 81646792-2576683487-2605776921-107194714 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1280331516 Contact: <sip:8432762180@172.22.12.3:5060> Call-Info: <sip:172.22.12.3:5060>;method="NOTIFY;Event=telephone-event;Duration=2000" Expires: 180 Allow-Events: telephone-event Content-Type: multipart/mixed;boundary=uniqueBoundary Mime-Version: 1.0 Content-Length: 1112

  8. SIP Messages • Trying 100 SIP/2.0 100 Trying Date: Wed, 28 Jul 2010 15:38:36 GMT From: "JEFFERSON MARTI" <sip:8432762180@172.22.12.3>;tag=B064C8D4-DF3 Allow-Events: presence Content-Length: 0 To: <sip:5620@172.22.5.10> Call-ID: 54322D8-999511DF-99D69AFE-2081E384@172.22.12.3 Via: SIP/2.0/UDP 172.22.12.3:5060;branch=z9hG4bK155691746 CSeq: 101 INVITE

  9. SIP Messages • Ringing 180 SIP/2.0 180 Ringing Date: Wed, 28 Jul 2010 15:38:36 GMT Call-Info: <sip:172.22.5.10:5060>;method="NOTIFY;Event=telephone-event;Duration=500" Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY From: "JEFFERSON MARTI" <sip:8432762180@172.22.12.3>;tag=B064C8D4-DF3 Allow-Events: presence P-Asserted-Identity: "Marty Jefferson" <sip:5620@172.22.5.10> Supported: X-cisco-srtp-fallback Supported: Geolocation Remote-Party-ID: "Marty Jefferson" <sip:5620@172.22.5.10>;party=called;screen=yes;privacy=off Content-Length: 0 To: <sip:5620@172.22.5.10>;tag=011176a9-9c00-4b03-b253-35ace3ee8112-19702291 Contact: <sip:5620@172.22.5.10:5060> Call-ID: 54322D8-999511DF-99D69AFE-2081E384@172.22.12.3 Via: SIP/2.0/UDP 172.22.12.3:5060;branch=z9hG4bK155691746 CSeq: 101 INVITE

  10. SIP Messages • Info INFO sip:5620@172.22.5.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.22.12.3:5060;branch=z9hG4bK1556ABF1 From: "JEFFERSON MARTI" <sip:8432762180@172.22.12.3>;tag=B064C8D4-DF3 To: <sip:5620@172.22.5.10>;tag=011176a9-9c00-4b03-b253-35ace3ee8112-19702291 Date: Wed, 28 Jul 2010 15:38:36 GMT Call-ID: 54322D8-999511DF-99D69AFE-2081E384@172.22.12.3 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Timestamp: 1280331516 CSeq: 102 INFO Contact: <sip:8432762180@172.22.12.3:5060> Remote-Party-ID: "JEFFERSON MARTI" <sip:8432762180@172.22.12.3>;party=calling;screen=yes;privacy=off Content-Type: multipart/mixed;boundary=uniqueBoundary Mime-Version: 1.0 Content-Length: 391

  11. SIP Messages • OK 200 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.22.5.10:5060;branch=z9hG4bK590fd0304b15ff From: "Marty Jefferson" <sip:7049435620@172.22.5.10>;tag=011176a9-9c00-4b03-b253-35ace3ee8112-19702287 To: <sip:918432762180@172.22.12.3>;tag=B05DBE70-718 Date: Wed, 28 Jul 2010 15:30:52 GMT Call-ID: 19c7a380-c5014d2c-58464b-a0516ac@172.22.5.10 CSeq: 101 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: <sip:918432762180@172.22.12.3>;party=called;screen=no;privacy=off Contact: <sip:918432762180@172.22.12.3:5060> Supported: replaces Call-Info: <sip:172.22.12.3:5060>;method="NOTIFY;Event=telephone-event;Duration=500" Supported: sdp-anat Server: Cisco-SIPGateway/IOS-12.x Session-Expires: 1800;refresher=uac Require: timer Content-Type: multipart/mixed;boundary=uniqueBoundary Mime-Version: 1.0 Content-Length: 670

  12. SIP Messages • ACK ACK sip:918432762180@172.22.12.3:5060 SIP/2.0 Date: Wed, 28 Jul 2010 15:38:37 GMT From: "Marty Jefferson" <sip:7049435620@172.22.5.10>;tag=011176a9-9c00-4b03-b253-35ace3ee8112-19702292 Allow-Events: presence, kpml Content-Length: 0 To: <sip:918432762180@172.22.12.3>;tag=B064D884-10C Call-ID: 2ef10200-c5014efd-584897-a0516ac@172.22.5.10 Via: SIP/2.0/UDP 172.22.5.10:5060;branch=z9hG4bK59121e2736c5b4 CSeq: 101 ACK Max-Forwards: 70

  13. SIP Messages • BYE BYE sip:5620@172.22.5.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.22.12.3:5060;branch=z9hG4bK1556CB3C From: "JEFFERSON MARTI" <sip:8432762180@172.22.12.3>;tag=B064C8D4-DF3 To: <sip:5620@172.22.5.10>;tag=011176a9-9c00-4b03-b253-35ace3ee8112-19702291 Date: Wed, 28 Jul 2010 15:38:36 GMT Call-ID: 54322D8-999511DF-99D69AFE-2081E384@172.22.12.3 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Timestamp: 1280331524 CSeq: 103 BYE Reason: Q.850;cause=16 Content-Type: multipart/mixed;boundary=uniqueBoundary Mime-Version: 1.0 Content-Length: 281

  14. Session Boarder Controller • The Cisco Unified Border Element (CUBE) facilitates simple and cost-effective connectivity between enterprise unified communications Session Initiation Protocol (SIP) trunks to the public-switched telephone network (PSTN). Designed to meet enterprise and service-provider Session Border Controller (SBC)

  15. IOS Gateway Configuration • SIP configuration is similar to H.323 • SIP uses Dial Peers dial-peer voice 2 voip destination-pattern 8... session protocol sipv2 session target ipv4:9.13.8.16 dtmf-relay rtp-nte

  16. IOS Gateway Configuration • SIP-UA section used if authentication is required sip-ua registrar ipv4:200.1.1.10 or registrar dns:csps.cisco.com authentication username xyz password xyz realm cisco.com

  17. IOS Gateway Configuration • Voice Service VoIP voice service voip allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip allow-connections h323 to h323

  18. UCM Configuration • Build CUBE and SIP Gateways as Trunks

  19. UCM Configuration • Build CUBE and SIP Gateways as Trunks

  20. Agenda • What is SIP? • Why SIP over T1/PRI? • SIP Trunking • Cisco SIP End points

  21. Why SIP over T1? • Increased Capacity • One 10MB SIP trunk equals 5.3 T1s • Added Features over T1 • SIP trunks can carry voice, video and application information • Quicker to Increase Capacity • Increasing capacity is just a software setting

  22. Why SIP over T1? • Increased Capacity • One 10MB SIP trunk equals 5.3 T1s • Added Features over T1 • SIP trunks can carry voice, video and application information • Quicker to Increase Capacity • Increasing capacity is just a software setting

  23. Agenda • What is SIP? • Why SIP over T1/PRI? • SIP Trunking • Cisco SIP End points

  24. SIP Trunking • SIP trunking is becoming more available from Telco Vendors like Triad Telecom. • Deployment models vary from MPLS connections to Dedicated circuits to Internet connections. • Cisco is using Internet connections with their Intercompany Media Exchange (IME)

  25. Agenda • What is SIP? • Why SIP over T1/PRI? • SIP Trunking • Cisco SIP End points

  26. Cisco SIP • 794X/6X phones are SCCP and SIP capable • 8961 and 99XX are SIP only devices • Cisco 2800/2900/3800/3900 IOS gateways can be SIP gateways • TelePresence Units • Unity, Unity Connection and Unity Express • MeetingPlace and MeetingPlace Express

  27. References • Cisco -SIP: The Next Step in Converged IP Communications http://www.cisco.com/en/US/partner/technologies/tk652/tk701/technologies_white_paper0900aecd80131325.html • VoIP-Info.org http://www.voip-info.org/wiki/view/SIP • Cisco Products & Services http://www.cisco.com/en/US/partner/products/sw/voicesw/index.html#~all-prod

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