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SIP
South Carolina Cisco User Group
Martin JeffersonIE 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
• InviteINVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.12.3:5060;branch=z9hG4bK155691746
Remote-Party-ID: "JEFFERSON MARTI" <sip:[email protected]>;party=calling;screen=yes;privacy=off
From: "JEFFERSON MARTI" <sip:[email protected]>;tag=B064C8D4-DF3
To: <sip:[email protected]>
Date: Wed, 28 Jul 2010 15:38:36 GMT
Call-ID: [email protected]
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:[email protected]: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
SIP Messages
• Trying 100SIP/2.0 100 Trying
Date: Wed, 28 Jul 2010 15:38:36 GMT
From: "JEFFERSON MARTI" <sip:[email protected]>;tag=B064C8D4-DF3
Allow-Events: presence
Content-Length: 0
To: <sip:[email protected]>
Call-ID: [email protected]
Via: SIP/2.0/UDP 172.22.12.3:5060;branch=z9hG4bK155691746
CSeq: 101 INVITE
SIP Messages
• Ringing 180SIP/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:[email protected]>;tag=B064C8D4-DF3
Allow-Events: presence
P-Asserted-Identity: "Marty Jefferson" <sip:[email protected]>
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Remote-Party-ID: "Marty Jefferson" <sip:[email protected]>;party=called;screen=yes;privacy=off
Content-Length: 0
To: <sip:[email protected]>;tag=011176a9-9c00-4b03-b253-35ace3ee8112-19702291
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
Via: SIP/2.0/UDP 172.22.12.3:5060;branch=z9hG4bK155691746
CSeq: 101 INVITE
SIP Messages
• InfoINFO sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.12.3:5060;branch=z9hG4bK1556ABF1
From: "JEFFERSON MARTI" <sip:[email protected]>;tag=B064C8D4-DF3
To: <sip:[email protected]>;tag=011176a9-9c00-4b03-b253-35ace3ee8112-19702291
Date: Wed, 28 Jul 2010 15:38:36 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1280331516
CSeq: 102 INFO
Contact: <sip:[email protected]:5060>
Remote-Party-ID: "JEFFERSON MARTI" <sip:[email protected]>;party=calling;screen=yes;privacy=off
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length: 391
SIP Messages
• OK 200SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.5.10:5060;branch=z9hG4bK590fd0304b15ff
From: "Marty Jefferson" <sip:[email protected]>;tag=011176a9-9c00-4b03-b253-35ace3ee8112-19702287
To: <sip:[email protected]>;tag=B05DBE70-718
Date: Wed, 28 Jul 2010 15:30:52 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
Contact: <sip:[email protected]: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
SIP Messages
• ACKACK sip:[email protected]:5060 SIP/2.0
Date: Wed, 28 Jul 2010 15:38:37 GMT
From: "Marty Jefferson" <sip:[email protected]>;tag=011176a9-9c00-4b03-b253-35ace3ee8112-19702292
Allow-Events: presence, kpml
Content-Length: 0
To: <sip:[email protected]>;tag=B064D884-10C
Call-ID: [email protected]
Via: SIP/2.0/UDP 172.22.5.10:5060;branch=z9hG4bK59121e2736c5b4
CSeq: 101 ACK
Max-Forwards: 70
SIP Messages
• BYEBYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.12.3:5060;branch=z9hG4bK1556CB3C
From: "JEFFERSON MARTI" <sip:[email protected]>;tag=B064C8D4-DF3
To: <sip:[email protected]>;tag=011176a9-9c00-4b03-b253-35ace3ee8112-19702291
Date: Wed, 28 Jul 2010 15:38:36 GMT
Call-ID: [email protected]
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
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 ipv4:9.13.8.16
dtmf-relay rtp-nte
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
IOS Gateway Configuration
• Voice Service VoIP
voice service voipallow-connections h323 to sipallow-connections sip to h323allow-connections sip to sipallow-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 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
![[MS-SIP]: Session Initiation Protocol ExtensionsMS-SIP].pdfSession Initiation Protocol Extensions SIP. . SIP message.](https://img.dokumen.tips/doc/110x75/5e7f8669844925290d6f8357/ms-sip-session-initiation-protocol-extensions-ms-sippdf-session-initiation.jpg)
