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Challenges and Opportunities Deploying in a SIP Environment

Challenges and Opportunities Deploying in a SIP Environment. Bob Cooper Chief Architect – Voice Portal bob@avaya.com +1 (803) 231-2715. SIP in general. Don’t think of SIP as just another telephony protocol It’s not the same as migrating from POTS to ISDN

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Challenges and Opportunities Deploying in a SIP Environment

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  1. Challenges and Opportunities Deploying in a SIP Environment Bob Cooper Chief Architect – Voice Portal bob@avaya.com +1 (803) 231-2715

  2. SIP in general Don’t think of SIP as just another telephony protocol It’s not the same as migrating from POTS to ISDN It’s more akin to migrating toward a web model for telephony Call routing, data passing, end-point addressing, security, … There are opportunities/advantages in migrating to SIP and There are some challenges along the way

  3. Simplified Enterprise View (PSTN and SIP)(don’t interpret too literally) SIP Deployment PSTN Deployment

  4. Application Selection/Addressing PSTN typically selects an application based on ANI (dialed number) DNIS (calling number…somewhat unreliable) Time of day Typical admin screen looks something like this

  5. Application Selection/Addressing SIP application selection Can be based on “To:” header or “Request URI” Request URI however can change along the way To: header can cause issues if call was re-directed (voicemail app)…in which case the “history-info:” header may need to be examined. Example sip:18001234567@company.com sip:app1@company.com?UID=1234&priority=urgent sip:support@company.com?app=application1

  6. Application Selection/Addressing SIP application selection Could specify startpage w/in SIP URI RFC 4240 Sip:dialog@company.com; voicexml=http://domain.com/startpage.vxml?account=1234

  7. Distributed Nature/Web Model PSTN VoiceXML Servers look like stations or ISDN trunks Traditional telecom equipment takes care of Load balancing – hunt groups across trunks/stations Failover conditions Busy conditions After hours conditions

  8. Distributed Nature/Web Model SIP (Web) Model – who takes care of… Routing Do all VXML servers run the same app or are apps assigned to specific VXML servers? Load balancing What type of load balancing scheme is used? Round robin, Least loaded (stateful), … Busy conditions Does a 486 BUSY get sent to the caller or does another server(s) get the INVITE Failure/No answer condition – fast timeout or OPTIONS Who solves this for you? Many of these are not traditional SIP Proxy functions?

  9. In Call Data Passing PSTN – extremely limited User to User Information (UUI) Typically limited to 128 Bytes SIP – very robust UUI URI parameters (in many of the headers) Dedicated SIP headers User Defined SIP headers MIME encoded data in SIP message body

  10. In Call Data Passing Examples RFC 4240 Sip:dialog@company.com; voicexml=http://domain.com/startpage.vxml?account=1234 URI parameters (applies to many headers) sip:app1@company.com?UID=1234&priority=urgent Defined by SIP Subject:”Hello World” Contact:”Bob Cooper”<sip:bob@avaya.com> User defined AccountOwner:”bob cooper” ScreenPop:<account>123456</account><CID>111</CID>…

  11. In Call Data Passing Questions to ask How does a VoiceXML/CCXML application get access (send and receive) this information? Are the methods and headers tied to a vendor’s platform or authoring environment? What does the application writer need to know about SIP to make use of this information?

  12. Asynchronous Events PSTN Not much to speak of SIP Info – sent anytime during a call In band REFER Out of band REFER VoiceXML 2.x was not designed to handle these types of events Look to CCXML and/or SCXML event driven protocols to make sure of these capabilities

  13. Security PSTN Somewhat secure by its physical nature Not that susceptible to denial of service attacks, … SIP Web model based on Mutual key exchange, certificate authorities, .. SIPS is secure SIP TLS for the SIP signaling path SRTP for the media path Several different methods of exchanging keys

  14. Feel free to contact me anytime Bob Cooper bob@avaya.com 803-231-2714

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