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? 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

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

SIP

SIP Messages Invite INVITE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK Remote-Party-ID: "JEFFERSON MARTI" ;party=calling;screen=yes;privacy=off From: "JEFFERSON MARTI" ;tag=B064C8D4-DF3 To: Date: Wed, 28 Jul :38:36 GMT Call-ID: Supported: timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 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: Contact: Call-Info: ;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

SIP Messages Trying 100 SIP/ Trying Date: Wed, 28 Jul :38:36 GMT From: "JEFFERSON MARTI" ;tag=B064C8D4-DF3 Allow-Events: presence Content-Length: 0 To: Call-ID: Via: SIP/2.0/UDP :5060;branch=z9hG4bK CSeq: 101 INVITE

SIP Messages Ringing 180 SIP/ Ringing Date: Wed, 28 Jul :38:36 GMT Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=500" Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY From: "JEFFERSON MARTI" ;tag=B064C8D4-DF3 Allow-Events: presence P-Asserted-Identity: "Marty Jefferson" Supported: X-cisco-srtp-fallback Supported: Geolocation Remote-Party-ID: "Marty Jefferson" ;party=called;screen=yes;privacy=off Content-Length: 0 To: ;tag=011176a9-9c00-4b03-b253-35ace3ee Contact: Call-ID: Via: SIP/2.0/UDP :5060;branch=z9hG4bK CSeq: 101 INVITE

SIP Messages Info INFO SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK1556ABF1 From: "JEFFERSON MARTI" ;tag=B064C8D4-DF3 To: ;tag=011176a9-9c00-4b03-b253-35ace3ee Date: Wed, 28 Jul :38:36 GMT Call-ID: User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Timestamp: CSeq: 102 INFO Contact: Remote-Party-ID: "JEFFERSON MARTI" ;party=calling;screen=yes;privacy=off Content-Type: multipart/mixed;boundary=uniqueBoundary Mime-Version: 1.0 Content-Length: 391

SIP Messages OK 200 SIP/ OK Via: SIP/2.0/UDP :5060;branch=z9hG4bK590fd0304b15ff From: "Marty Jefferson" ;tag=011176a9-9c00-4b03-b253-35ace3ee To: ;tag=B05DBE Date: Wed, 28 Jul :30:52 GMT Call-ID: CSeq: 101 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Contact: Supported: replaces Call-Info: ;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

SIP Messages ACK ACK SIP/2.0 Date: Wed, 28 Jul :38:37 GMT From: "Marty Jefferson" ;tag=011176a9-9c00-4b03-b253-35ace3ee Allow-Events: presence, kpml Content-Length: 0 To: ;tag=B064D884-10C Call-ID: Via: SIP/2.0/UDP :5060;branch=z9hG4bK59121e2736c5b4 CSeq: 101 ACK Max-Forwards: 70

SIP Messages BYE BYE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK1556CB3C From: "JEFFERSON MARTI" ;tag=B064C8D4-DF3 To: ;tag=011176a9-9c00-4b03-b253-35ace3ee Date: Wed, 28 Jul :38:36 GMT Call-ID: User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Timestamp: CSeq: 103 BYE Reason: Q.850;cause=16 Content-Type: multipart/mixed;boundary=uniqueBoundary Mime-Version: 1.0 Content-Length: 281

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)

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: dtmf-relay rtp-nte

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

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

UCM Configuration Build CUBE and SIP Gateways as Trunks

UCM Configuration Build CUBE and SIP Gateways as Trunks

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

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

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

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

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)

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

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

References Cisco -SIP: The Next Step in Converged IP Communications k701/technologies_white_paper0900aecd html k701/technologies_white_paper0900aecd html VoIP-Info.org Cisco Products & Services /voicesw/index.html#~all-prod /voicesw/index.html#~all-prod