© 2007 avaya inc. all rights reserved. challenges and opportunities deploying in a sip environment...

© 2007 Avaya Inc. All rights reserved.

Challenges and Opportunities Deploying in a SIP Environment Challenges and Opportunities

Deploying in a SIP Environment

Bob CooperChief Architect – Voice Portal

[email protected]+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

– 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 SIPand

There are some challenges along the way


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

SIP Deployment

PSTN Deployment

PBXCO

SBCServiceProvider

Proxy

GatewayCO

Node 1

Node 2

Node n

App ServerApp Server


Admin


App ServerASR/TTS

Node 1

Node 2

Node n



Admin


App ServerASR/TTS

VoiceXML/CCXML Engines

VoiceXML/CCXML Engines


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

DNIS Start Page Attributes

800-123-4567 http://example.com/bob.vxml Lang, ….

888-123-4568 https://example.com/bob.jsp Lang, …

… … …



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:[email protected]

– sip:[email protected]?UID=1234&priority=urgent

– sip:[email protected]?app=application1



SIP application selection

– Could specify startpage w/in SIP URI

– RFC 4240

– Sip:[email protected]; voicexml=http://domain.com/startpage.vxml?account=1234

URI Start Page Attributes

[email protected] http://company.com/bob.vxml Lang, ….

.*company.com https://company.com/generic.jsp Lang, …


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


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?


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



Examples– RFC 4240

• Sip:[email protected]; voicexml=http://domain.com/startpage.vxml?account=1234

– URI parameters (applies to many headers)• sip:[email protected]?UID=1234&priority=urgent

– Defined by SIP• Subject:”Hello World”

• Contact:”Bob Cooper”<sip:[email protected]>

– User defined• AccountOwner:”bob cooper”

• ScreenPop:<account>123456</account><CID>111</CID>…



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?


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


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


Feel free to contact me anytime

Bob Cooper

[email protected]

803-231-2714

© 2007 avaya inc. all rights reserved. challenges and opportunities deploying in a sip environment...

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