SIP Application Note

SIP Application Note
Installation and Reference Guide
Customer Interaction Center®

Enterprise Interaction Center®
Communité®
Version 2.4
Last updated 12/1/2005

(See Change Log for summary of change made to this document since GA.)
Always check for a newer version of this document!

Application Notes: http://www.inin.com/support/cic/23/telephony/documentation.asp
Abstract
This document contains instructions for installing and configuring SIP functionality on your 2.4 CIC or
EIC Server. Please note: This is a work in progress. IC 2.4 functionality will replace existing
IC 2.3 functionality.
This document applies to one or more Interactive Intelligence and/or Vonexus products. Vonexus is a
wholly owned subsidiary of Interactive Intelligence.
Copyright and Trademark Information
©1994 – 2005 Interactive Intelligence Inc./ Vonexus Inc. All rights reserved. Vonexus is a wholly-owned subsidiary
of Interactive Intelligence Inc. Interactive Intelligence®, Interaction Center Platform®, Communité®, Enterprise
Interaction Center®, Interactive Intelligence Customer Interaction Center®, e-FAQ®, e-FAQ Knowledge Manager,
Interaction Dialer®, Interaction Director®, Interaction Marquee, Interaction Recorder®, Interaction SIP Proxy,
Interaction Supervisor, Interaction Tracker, Mobilité®, Vocalité®, Interaction Administrator®, Interaction
Attendant®, Interaction Client®, Interaction Designer®, Interaction Fax Viewer, Interaction FAQ, Interaction
Melder, Interaction Screen Recorder, Interaction Scripter®, Interaction Server, Wireless Interaction Client,
InteractiveLease®, and the “Spirograph” logo design® are all trademarks or registered trademarks of Interactive
Intelligence Inc.
veryPDF is Copyright © 2000-2005 by veryPDF, Inc. Other brand and/or product names referenced in this
document are the trademarks or registered trademarks of their respective companies.
Interactive Intelligence Inc.
7601 Interactive Way
Indianapolis, Indiana 46278
Telephone/Fax (317) 872-3000
www.ININ.com
Vonexus
7601 Interactive Way
Indianapolis, Indiana 46278
Telephone/Fax (888) 817-5904
www.vonexus.com
DISCLAIMER
INTERACTIVE INTELLIGENCE (INTERACTIVE) HAS NO RESPONSIBILITY UNDER WARRANTY, INDEMNIFICATION OR
OTHERWISE, FOR MODIFICATION OR CUSTOMIZATION OF ANY INTERACTIVE SOFTWARE BY INTERACTIVE,
CUSTOMER OR ANY THIRD PARTY EVEN IF SUCH CUSTOMIZATION AND/OR MODIFICATION IS DONE USING
INTERACTIVE TOOLS, TRAINING OR METHODS DOCUMENTED BY INTERACTIVE.
Interaction Center Platform Statement

This document describes Interaction Center (IC) features that may not be available in your IC product.
Several products are based on the IC platform, and some features are disabled in some products.
Three products are based on the IC platform:
• Customer Interaction Center (CIC)
• Enterprise Interaction Center (EIC)
• Communité
While all of these products share a common feature set, this document is intended for use with all IC
products, and some of the described features may not be available in your product.
How do I know if I have a documented feature?

Here are some indications that the documented feature is not available in your version:
• The menu, menu item, or button that accesses the feature appears grayed-out.
• One or more options or fields in a dialog box appear grayed-out.
• The feature is not selectable from a list of options.
If you have questions about feature availability, contact your vendor regarding the feature set
available in your version of this product.
SIP Application Note 2 of 129 ©2005 Interactive Intelligence, Inc.

Table of Contents
1 Change Log ................................................................................................................... 10
2 Where can I get information? ....................................................................................... 11
2.1 Interactive Intelligence Web Site.................................................................................11
2.2 Third Party Component Certification ............................................................................11
2.3 Software Versions and Upgrades .................................................................................11
3 What’s New .................................................................................................................. 12
4 Known Issues ............................................................................................................... 12
4.1 Known Issues with Interaction Center Products .............................................................12
4.2 Known Issues with 3rd Party Products ..........................................................................12
4.2.1 Known Issues with 3rd Party Products (General) ......................................................12
4.2.2 Known Issues with 3rd Party Products (Microsoft).....................................................13
4.2.3 Known Issues with 3rd Party Products (Cisco)..........................................................14
4.2.4 Known Issues with 3rd Party Products (ActionTec)....................................................14
5 Glossary of Terms ......................................................................................................... 14
6 Introduction ................................................................................................................. 15
6.1 Available SIP-Related Application Notes .......................................................................15
6.2 Standards................................................................................................................15
6.2.1 Other Companies................................................................................................15
6.2.2 What is an RFC...................................................................................................15
6.2.3 SIP Standards ....................................................................................................16
6.2.4 Why has RFC 2543 been replaced with RFC 3261? ...................................................17
6.2.5 IP Address and Ports ...........................................................................................17
6.3 SIP Q&A ..................................................................................................................18
6.4 Implementation Overview Diagrams ............................................................................21
6.4.1 SIP Hardware Approach Overview .........................................................................21
7 When is a SIP Proxy Needed? ....................................................................................... 21
7.1 SIP Message Routing.................................................................................................22
7.2 Phone Specific Routing ..............................................................................................22
7.2.1 When is a SIP proxy needed for the SIP phones I’m using? .......................................22
7.2.2 Gateway Specific Routing.....................................................................................23
7.2.3 When is a SIP proxy needed for the gateways I’m using? .........................................23
8 Telephony Connectivity Options with the Interaction Center ........................................ 23
8.1 PSTN Connectivity Options .........................................................................................24
8.2 Phone Options..........................................................................................................26
8.3 Remote Survivability and Emergency Dialing ................................................................27
8.3.1 Cisco’s NON-SIP SRST (Survivable Remote Site Telephony) ......................................27

8.3.2 Cisco’s SIP SRST (Survivable Remote Site Telephony)..............................................28
8.3.3 Interactive Intelligence’s Remote Survivability using SIP ..........................................29
8.3.4 Emergency (911) dialing using SIP........................................................................29
9 Audio Path and Resource Usage ................................................................................... 30
9.1 Dynamic Audio .........................................................................................................30
9.1.1 Remote Sites Without Remote Gateways ................................................................30
9.1.2 Remote Sites with Remote Gateways.....................................................................30
9.2 Conference Resources ...............................................................................................30
9.3 IP Resources ............................................................................................................31
9.4 Bandwidth Usage ......................................................................................................31
9.5 Sample Systems.......................................................................................................31
9.6 Putback Transfers (similar to ISDN RLT “Release Link Transfers”) ....................................32
9.7 External Audio Path...................................................................................................33
9.7.1 External Audio Path Resource Usage......................................................................33
9.7.2 External Audio Codec Selection (Device A to IC)......................................................34
9.7.3 External Audio Codec Selection (IC to Device B)......................................................34
10 Voice Issues on Networks.......................................................................................... 35
10.1 Quality of Service (QoS) .........................................................................................35
10.1.1 Layer 3 IP Header Byte.....................................................................................35
10.1.2 Layer 2 Byte (802.1p/Q)...................................................................................36
10.1.3 Notes about QoS and the Interaction Client. ........................................................36
10.2 Echo ....................................................................................................................37
10.3 RTCP Sender Reports .............................................................................................37
11 Security ..................................................................................................................... 38
11.1.1 Security Alert ..................................................................................................38
11.1.2 IC features .....................................................................................................38
12 Firewalls and NAT ...................................................................................................... 39
12.1 Cisco Firewall Information .......................................................................................39
12.2 VPN .....................................................................................................................39
13 Notes About User and Station Extensions .................................................................. 40
14 Inbound Logic............................................................................................................ 40
14.1 Diversion..............................................................................................................40
14.2 Inbound Calls, including DID ...................................................................................40
14.3 SIP Info Message...................................................................................................43
15 Outbound Logic.......................................................................................................... 43
16 Platforms ................................................................................................................... 45
16.1 Platform Combinations and Supported Status ............................................................45
16.2 Platform Comparison ..............................................................................................45

17 Installing and Configuring AudioCodes Boards .......................................................... 48
17.1 Important Notes and Restrictions .............................................................................48
17.2 Servers ................................................................................................................48
17.3 Known Issues........................................................................................................48
17.4 AudioCodes with Dialogic ........................................................................................49
17.4.1 Physical Placement of Boards ............................................................................49
17.4.2 Dialogic’s 3rd Party Board Configuration...............................................................49
17.4.3 Turning off Secondary Clock Master....................................................................49
17.5 AudioCodes with Aculab..........................................................................................49
17.6 a-law and mu-law ..................................................................................................49
17.7 Prerequisites .........................................................................................................50
17.8 AudioCodes PCI Driver............................................................................................50
17.9 Configuring the AudioCodes Boards with Interaction Administrator................................53
18 Installing and Configuring Intel HMP Software Solution............................................ 55
18.1 Important Notes and Restrictions .............................................................................55
18.2 Servers ................................................................................................................55
18.3 Densities ..............................................................................................................55
18.4 Vendor Software....................................................................................................56
18.5 Configuring your HMP system. .................................................................................57
18.5.1 Service Setting................................................................................................57
18.5.2 QoS Setting ....................................................................................................57
18.5.3 IP addresses ...................................................................................................57
18.5.4 Timers ...........................................................................................................57
18.6 Configuring RTP Dynamic Port Range........................................................................58
18.7 Known IC Issues with HMP ......................................................................................59
18.8 Known HMP Issues.................................................................................................59
18.9 HMP Limitations.....................................................................................................61
19 Creating and Modifying SIP Lines in Interaction Administrator ................................. 61
19.1 Line Configurations not exposed through Interaction Administrator ...............................61
19.2 Creating A SIP Line ................................................................................................61
19.3 Line Configuration: Line Page .................................................................................62
19.4 Line Configuration: Audio Page ...............................................................................64
19.5 Line Configuration: Transport Page..........................................................................66
19.6 Line Configuration: Session Page ............................................................................67
19.7 Line Configuration: Authentication Page ...................................................................69
19.8 Line Configuration: Compression Page .....................................................................70
19.9 Line Configuration: Proxy Page ...............................................................................72
19.10 Line Configuration: Registrar Page ..........................................................................73

19.11 Line Configuration: Access Page..............................................................................74
19.12 Line Configuration: Region Page .............................................................................75
19.13 Line Configuration: Call Putback Page ......................................................................75
20 Defining Global Configurations for SIP Stations......................................................... 77
20.1 Global SIP Station Configurations not exposed through Interaction Administrator ............77
20.2 Global Station Configuration: Addresses Page ...........................................................78
20.2.1 Notes on “Dynamically Updated Contact Addresses” and the audio-enabled client. ....78
20.3 Global Station Configuration: Audio Page .................................................................79
20.4 Global Station Configuration: Transport Page ............................................................80
20.5 Global Station Configuration: Session Page...............................................................81
20.6 Global Station Configuration: Authentication Page .....................................................83
20.7 Global Station Configuration: Compression Page .......................................................84
20.8 Global Station Configuration: Phone Page.................................................................85
21 Creating and Configuring SIP stations in Interaction Administrator .......................... 86
21.1 Creating A SIP Station ............................................................................................86
21.2 SIP Station Configurations not exposed through Interaction Administrator .....................87
21.3 Station Configuration: Addresses Page.....................................................................87
21.3.1 Identification SIP Address Page .........................................................................87
21.3.2 Connection SIP Address Page ............................................................................89
21.4 Station Configuration: Audio Page ...........................................................................91
21.5 Station Configuration: Transport Page .....................................................................92
21.6 Station Configuration: Session Page ........................................................................93
21.7 Station Configuration: Authentication Page ...............................................................94
21.8 Station Configuration: Compression Page .................................................................95
21.9 Station Configuration: Phone Page ..........................................................................96
21.10 Station Configuration: General Page ........................................................................96
21.11 Station Configuration: Appearances.........................................................................98
21.12 Station Configuration: Region .................................................................................99
21.13 Station Configuration: Station Options Tab ...............................................................99
22 SIP Telephony Parameters in Interaction Administrator ......................................... 100
22.1 Server Configuration: SIP Telephony Parameters Page ............................................. 100
22.2 Configuring the Message Button for Voicemail Retrieval............................................. 101
22.2.1 Setup .......................................................................................................... 101
22.2.2 Vendor Specific ............................................................................................. 102
22.3 Configuring Voice Mail for Non-Managed Phones (SIP Diversion) ................................. 102
22.3.1 Logic ........................................................................................................... 103
22.3.2 Setup .......................................................................................................... 104
22.3.3 Vendor Specific ............................................................................................. 104

22.4 Configuring the Managed Phone Shortcut ................................................................ 105
22.4.1 Setup .......................................................................................................... 105
22.5 Configuring Message Waiting Indicators (MWI)......................................................... 105
22.5.1 Vendor Specific ............................................................................................. 106
23 Dial Plan Basics for SIP............................................................................................ 106
23.1 Dial Plan General Info........................................................................................... 106
23.2 Dial Plan Verification and Testing ........................................................................... 109
24 Gateway/Proxy Configuration ................................................................................. 110
24.1 Dial Plan: Configuring Gateway Selection ................................................................ 111
24.2 Dial Plan: Configuration of Displayed Numbers ........................................................ 112
24.2.1 Example 1 .................................................................................................... 113
24.2.2 Example 2 .................................................................................................... 113
24.3 Multiple Gateway Configuration.............................................................................. 113
24.3.1 Detecting Gateway Failure and/or Congestion .................................................... 114
24.3.2 Configuring Gateway Selection by using an External Proxy................................... 114
24.3.3 Configuring Gateway Selection by DialPlan ........................................................ 114
25 Call Analysis ............................................................................................................ 114
25.1 Call Analysis (over traditional connections) .............................................................. 114
25.2 Call Analysis (over IP) .......................................................................................... 114
25.3 Call Analysis (over IP) with Interaction Dialer .......................................................... 115
26 Fax Configuration .................................................................................................... 115
26.1 Availability.......................................................................................................... 115
26.2 Fax Detection...................................................................................................... 116
26.3 Scenarios ........................................................................................................... 116
26.3.1 Inbound Scenario .......................................................................................... 116
26.3.2 Outbound Scenario ........................................................................................ 116
26.4 IC Server Configuration ........................................................................................ 117
26.5 Gateway Configurations........................................................................................ 117
26.5.1 Cisco ........................................................................................................... 117
26.6 Potential Issues ................................................................................................... 117
27 Modem Configuration............................................................................................... 118
28 Tie Line and Multi-site Configuration ....................................................................... 118
29 Switchover Configuration ........................................................................................ 118
29.1 Switchover Component......................................................................................... 118
29.2 Audiocodes Configurations .................................................................................... 119
29.3 Station Configurations .......................................................................................... 119
29.4 Switchover in a WAN Environment ......................................................................... 119
30 Interaction Client Configuration .............................................................................. 119

30.1 Associating the Interaction Client with a Station ....................................................... 119
30.2 Configuring the Interaction Client for Audio ............................................................. 120
30.2.1 Special Messenger Considerations for SIP Enabled Interaction Client ..................... 121
30.2.2 Special Server Considerations for SIP Enabled Interaction Client........................... 121
30.3 Monitoring SIP Line Activity with the Interaction Client .............................................. 121
31 Phone Services ........................................................................................................ 122
32 Server Parameters................................................................................................... 122
33 Troubleshooting....................................................................................................... 123
33.1 Viewing Call Information ....................................................................................... 123
33.2 Tracing .............................................................................................................. 123
33.3 No Audio Problems............................................................................................... 124
33.4 Echo .................................................................................................................. 124
33.5 Audio Quality Problems......................................................................................... 124
33.6 DTMF Problems ................................................................................................... 125
33.6.1 IVR DTMF Recognition Problem ........................................................................ 125
33.6.2 No IVR, Plays, or records ................................................................................ 125
33.6.3 DTMF from Managed Phone not being recognized by remote system ..................... 125
33.7 Miscellaneous...................................................................................................... 125
33.7.1 Selecting hold on the Interaction client puts the call in Held, put the IP phone still
shows connected. ......................................................................................................... 125
33.7.2 All incoming calls going immediately to held state .............................................. 125
33.7.3 External Call made from SIP phone hears IVR rather than making the intended call. 126
33.7.4 Internal Call made from SIP phone is placed correctly, but does not show up on client.
126
33.7.5 Calls made from SIP phones do not show on Line Details Page ............................. 126
33.7.6 Phone rings when I use the MakeCall button in the Interaction Client .................... 126
33.7.7 Managed station not ringing ............................................................................ 126
33.7.8 Managed station not ringing ............................................................................ 126
33.7.9 Message Button playing the main menu ............................................................ 127
33.7.10 Microsoft Messenger window pops for every incoming call with using the SIP enabled
Interaction Client .......................................................................................................... 127
33.7.11 “Station Not Reached” error when making calls from the Interaction Client (when using
a SIP station) ............................................................................................................... 127
33.7.12 SIP Address has a “^” in it.............................................................................. 127
33.7.13 After hitting the Pickup or MakeCall buttons on my Interaction Client, I still must pick
up the handset to answer the call.................................................................................... 127
34 Tools........................................................................................................................ 127
34.1 Command Line Tools ............................................................................................ 127
34.2 Coder Bandwidth Usage ........................................................................................ 128
34.3 Speakeasy .......................................................................................................... 128

34.4 RTP Audio Monitor and Analysis Guide .................................................................... 128
35 Index ....................................................................................................................... 129

1 Change Log
The following changes have been made since this document was printed.
Authors: If you are making a change to this document, update the cover page date to match the date
of your latest changes.
Change Date
New for 2.4 12/1/05

2 Where can I get information?
The following section list sources of information for configuring, installing and troubleshooting SIP-
enabled Interaction Center products. It also lists resources for configuring, installing and
troubleshooting 3rd party devices that may be used with our SIP-enabled products.
2.1 Interactive Intelligence Web Site

Visit Interaction Intelligence’s General support web site for more information and more documents
regarding supported platforms and supported releases.
Visit Interaction Intelligence’s SIP support web site for the latest information on our SIP-enabled
products and supporting documentation.
Visit Interactive Intelligences Telephony Platform support web site for detailed information regarding
the telephony platforms for our product lines. The following documents are available for download
from this location
SIP Application Note
This document contains information focused on the operation and configuration of Interactive
Intelligence’s SIP-enabled products. These products include Customer Interaction Center (CIC),
Enterprise Interaction Center (EIC) and Communité.
SIP 3rd Party Component Feature Matrix
This document contains information about both certified and uncertified SIP devices, and what
features these devices have. Uncertified devices have been tested by Interactive Intelligence and
where found to have certain deficiencies or lack of market demand which kept them off our certified
list. Uncertified devices are listed for feature comparison only and should be used at your
own risk. You might be asked to remove an uncertified device from the network if support
is needed.
SIP 3rd Party Component Application Note
This document contains Interaction Center specific configuration information for both certified and
uncertified SIP devices. Uncertified devices are listed for information only and should be used
at your own risk.
2.2 Third Party Component Certification

Interactive Intelligence has certified SIP devices from 3rd party manufacturers that are available for
use with our Interaction Center and Communité products. These devices have been tested and known
to work well with our products. The complete list of certified devices is available in the SIP 3rd Party
Component Feature Matrix. This document is a matrix of features so each device can be compared
with others.
2.3 Software Versions and Upgrades

Get the latest versions of software.
• Interactive Intelligence: Hot fixes for each release are on the hotfix web site and listed
below. You must publish the new handlers that are in IC service releases (the handlers are not
automatically published when you install service releases).
• AudioCodes: If you are using AudioCodes IP boards, you should install the board and it’s
related files as instructed in section 17 “Installing and Configuring AudioCodes Boards”.
• Intel/Dialogic Software (HMP): If you are using Intel/Dialogic Host Media Processing
(HMP) Software, you should install as instructed in 18 “Installing and Configuring Intel HMP
Software Solution”
SIP Application Note 11 of 129© 2004 Interactive Intelligence, Inc.

3 What’s New
2.4 is a huge release for SIP enhancements. Below are a few. See the complete 2.4 documentation
for the complete list.
• Support for RFC 3261 (3261 replaced RFC 2543).
• SIP over TCP (now both UDP and TCP SIP connections are available).
• Access control lists in the line configuration for increased security.
• Interaction Media Server.
• Regionalization (a way to group your devices for codec and dial plan configuration).
• Support for AudioCodes firmware 4.6 for the IPM 260 IP boards.
• Support for HMP version 1.3.
• Increased scalability. HMP 1.3 has double the scalability vs. HMP 1.1. Using the Interaction
Media Server drastically increases the number of calls that can be processed.
4 Known Issues
4.1 Known Issues with Interaction Center Products

The following section identifies known issues with SIP-enabled Interaction Center products. It also
describes a workaround if one exists, it lists the release in which the issue was discovered, it lists the
release in which the issue was fixed and, finally, it lists any applicable hotfixes that may be applied to
older releases to fix the issue.
Issue Double digit detected
Description Double digit detected on last digit of an internal call. Also, the last digit dialed might be taken as
the first digit when in the auto attendant.
Workaround None
Affected IC 2.4 GA
Fixed TBA
Hotfixes None
4.2 Known Issues with 3rd Party Products

The following section identifies any known issues with 3rd party products used in conjunction with our
SIP-enabled Interaction Center products.
4.2.1 Known Issues with 3rd Party Products (General)
Issue Some SIP devices do not support delayed media
Product General
Description Some SIP devices do not support delayed media which is used, by default, for outbound calls
from the Interaction Center.
Symptom Calls initiated from an IC client fail to connect, are immediately disconnected or have one-way

Issue Some SIP devices do not support delayed media
audio.
Workaround Change the default behavior from delayed media to normal media on the Interaction Center by
checking the Disable Delayed Media checkbox on the SIP line, SIP station, and/or the global SIP
station containers.
Affected IC 2.4 GA
Fixed A fix is not necessary. Most devices support delayed media
Hotfixes Not applicable.
4.2.2 Known Issues with 3rd Party Products (Microsoft)
Issue Microsoft RTC DLL will not receive the first 8-12 seconds of audio.
Product Microsoft RTC DLL, which is used by the audio-enable Interaction Client (i.e. the Interaction Client
using the /mssipaudio flag)
Description When using the IC client as a soft phone, the first 8 to 12 seconds of audio (directed to the
Microsoft RTC DLL) is ignored by the Microsoft RTC. This occurs when the call is setup or an
advanced feature is performed on it, such as a record or hold operation. The IC client soft phone
uses Microsoft’s RTC library, which has a delay when moving audio streams.
Symptom 8 to 12 seconds of audio (to the Microsoft RTC DLL) is missing when a call is setup or an
advanced feature is performed, such as a record or hold operation.
Workaround Change the Audio Flow configuration from Dynamic to Always In on the SIP station and/or SIP
global station containers.
Affected RTC 1.1
Fixed
Hotfixes Not applicable
Issue Microsoft RTC DLL causes audio delay on long phone calls.
Description When using the IC client as a soft phone, audio (to the Microsoft RTC DLL) can be delayed by 2 to
3 seconds on long phone calls.
Symptom As the call progresses, the clock drift on the RTP packets drifts, and the TRC RTP stack delays
audio to the USB device.
Workaround Change the Voice Activation Detection (VAD) checkbox to checked (ON). in the SIP station and/or
SIP global station containers. Note that using VAD could cause audio problems (first spoken
word could be truncated.
Affected RTC 1.1
Fixed
Issue Microsoft RTC DLL changes port number on every audio change.
Description When using the IC client as a soft phone, audio (to the Microsoft RTC DLL) changes ports when

the IC server instructs the RTC DLL to send its audio to a different port.
Symptom Interactive Intelligence SIP loop detect will disconnect the call.
Workaround Change the Audio Flow configuration from Dynamic to Always In on the SIP station and/or SIP
global station containers.
Affected RTC 1.1
Fixed RTC 1.2
4.2.3 Known Issues with 3rd Party Products (Cisco)
Issue Cisco ATA 18x device ddoes not respond to SIP OPTIONS messages.
Product Cisco ATA 18x 3.1 firmware (all firmware versions).
Description Cisco ATA 18x devices does not respond to SIP OPTIONS messages.
Symptom Calls for users on Cisco SIP phones can be disconnected.
Workaround Uncheck the Use Session Timers checkbox in the SIP station and/or SIP global station container.
Affected IC 2.3
Fixed
4.2.4 Known Issues with 3rd Party Products (ActionTec)
Issue IC Client soft phone users hear buzz when answering calls
Product ActionTec analog to USB adapter
Description IC client soft phone users that use an ActionTec device may experience a buzz heard by the
remote caller when the phone plugged into the ActionTec goes off-hook.
Symptom Remote caller hears buzz when ActionTec user goes off-hook.
Workaround None
Affected Not applicable
Fixed Not applicable
5 Glossary of Terms
Term Description
Managed SIP phone that is configured as a SIP station in the Interaction Center. A SIP station is configured
phone in the Stations page of Interaction Administrator
Unmanaged SIP phone that is unknown to the Interaction Center.

phone
6 Introduction
With SIP (Session Initiation Protocol) being the emerging standard now used for call routing, state
functions and control within IP Networks, Interactive Intelligence now offers interoperability with SIP-
based solutions. As an open software solution, the Interactive Intelligence product line was designed
as a flexible and affordable alternative to traditional telecom solutions. With support for SIP,
Interactive Intelligence is excited to leverage it’s proven Interaction Center Platform to contact
centers, enterprises, e-businesses and service providers that wish to take advantage of the benefits a
converged network provides.
Although SIP-based soft-switches provide an excellent answer for next generation call transport over
packet networks, they still lack the compelling applications that will drive the level of acceptance that
their unique offerings strive to achieve. For example, capabilities as simple as voice mail and music
on-hold are not available. Interaction Center Platform answers this shortcoming by not only adding
these, but many more.
6.1 Available SIP-Related Application Notes

• Intel Hardware Application Note. How to install Intel Hardware Releases 5.1.1, 6.0.
• SIP Application Note. How to configure AudioCodes, Intel/Dialogic and Interaction Center for
SIP. (this guide)
• SIP Topology and Call Flows Application Note. High level view of the topologies and flows of a
SIP enabled network.
• SIP 3rd Party Component Application Note. How to configure different proxies, gateways, and
phones.
6.2 Standards
6.2.1 Other Companies

Interactive Intelligence has chosen SIP as its VoIP (Voice Over IP) solution for communication to
phones and gateways. It seems we are not alone. Many other companies understand the benefits of
a standards-based VoIP protocol and are aligning their products around a common vision.
Microsoft has publicly endorsed SIP as its protocol of choice for its Real Time Communications
initiative. Currently, Windows Messenger uses SIP for voice, instant messaging and presence and
Windows Live Communications Server is a SIP proxy, registrar, and load balancer.
Cisco has SIP enabled much of its product line. This includes not only some of their IP phones, but
also includes firewalls, gateways, and proxy servers. For a complete list see Cisco SIP-enabled
products web site.
Interactive Intelligence has been deploying SIP as a standard option since our 2.2 Interaction Center
release in June of 2002. Currently, many Interactive Intelligence customers use the Interaction
Center with phones and gateways from many compliant 3rd party manufactures.
6.2.2 What is an RFC

The Requests for Comments (RFC) document series is a set of technical and organizational notes
about the Internet. Memos in the RFC series discuss many aspects of computer networking, including
protocols, procedures, programs, and concepts, as well as meeting notes, opinions, and sometimes
humor. The official specification documents of the Internet Protocol suite that are defined by the
Internet Engineering Task Force (IETF) and the Internet Engineering Steering Group (IESG ) are
recorded and published as standards track RFCs. As a result, the RFC publication process plays an

important role in the Internet standards process. RFCs must first be published as Internet Drafts. The
Internet Standards Process itself is described in an RFC
6.2.3 SIP Standards

SIP standards are evolving quickly, and the Interaction Center continues to adhere to the
specifications for this emerging open standard. Below are the specifications implemented in
Interaction Center. These will continue to change as the drafts solidify and become standards.
RFC Standards Description
RFC 3261 (replaces RFC 2543) Session Initiation Protocol

TCP mandatory
Via branch id replaces call leg id as the transaction id
Url comparison rules were relaxed
Supported header for extensions advertising
New route/record-route simplification
RFC 2543bis04 Session Initiation Protocol.
RFC 2327 Session Description Protocol Description of the session within the SIP messages
RFC 2617 Basic and Digest Access Only Digest Access Authentication is supported. Basic
Authentication Access has been deprecated by RFC3261 (SIP) and is
not supported.
RFC 3265 SIP Specific Event Notification This document describes an extension to SIP. The
purpose of this extension is to provide an extensible
framework by which SIP nodes can request notification
from remote nodes indicating that certain events have
occurred. (SUBSCRIBE/NOTIFY)
RFC 3515 SIP REFER Method This document describes the REFER method that’s
most commonly used for call transfer operations.
RFC 3389 RTP Payload for Comfort Noise (CN) This document describes a Real-time Transport
Protocol (RTP) payload format for transporting comfort
noise (CN). The CN payload type is primarily for use
with audio codecs that do not support comfort noise as
part of the codec itself such as ITU-T
Recommendations G.711 and G.726.
RFC 3428 SIP Extension for Instant Messaging This document describes the MESSAGE method, an
extension to the SIP that allows the transfer of Instant
Messages. Interaction Center only supports
generating MESSAGE requests.
RFC 3581 An Extension to SIP for Symmetric This document describes symmetric response routing.
Response Routing When used with UDP, responses to requests are
returned to the source address the request came from,
and to the port written into the topmost Via header
field value of the request. This helps UDP packets
traverse firewalls.
RFC 3842 A Message Summary and Message This document describes a SIP event package to carry
Waiting Indication Event Package for SIP message waiting status and message summaries from
a messaging system. MWI using SUBSCRIBE/NOTIFY.
RFC 3891 SIP Replaces Header This document describes the Replaces header that can
be used to direct a called party to replace an existing
session with this one. It’s commonly used with REFER
for consult transfers.

RFC Standards Description
RFC 3911 SIP JOIN Header This document describes the JOIN header. Interaction
Center accepts JOIN requests as a mechanism to setup
conferences from SIP phones.
RFC Drafts Description
draft-levy-sip-diversion-03 Voicemail for unmanaged phones (uses Diversion/CC-

Diversion)
draft-ietf-sip-service-examples-03 Hold
draft-ietf-sipping-realtimefax-01.txt This document describes T.38 faxing call flows using

SIP.
6.2.4 Why has RFC 2543 been replaced with RFC 3261?
RFC 2543 has been deprecated and has been replaced by RFC3261. The new RFC clarifies and
resolves issues and mistakes made in RFC 2543. In addition to clarification, the text is much easier to
read and introduces a model for stateful transactions. On the technical side there have been a number
of changes including:
• TLS and S/MIME have been introduced and PGP removed
• Loose routing has been added to record routing which greatly increase the utility of record
routing
• Server location can be done with NAPTR records
• The syntax has been converted to ABNF and so can be checked automatically by standard tools
Due to these changes and others, the new RFC document is also “Standards Track” (The same rung on
the IETF standards ladder as RFC 2543.) It is proposed that once the new RFC has had time to be
implemented and tested, work will be carried out to advance SIP to “Proposed Standard” via a new
RFC.
RFC3261 is completely backward compatible with RFC2543.
6.2.5 IP Address and Ports
The following table shows the significant IP address and port numbers used by the Interaction Center
application.
Protocol IP Address Used Port Number Used
SIP over UDP IP Address of the system’s NIC Default is 5060. This is configurable on the
(Network Interface Card) SIP Line container in Interaction
Administrator.
SIP over TCP IP Address of the system’s NIC Default is 5060. This is configurable on the
(Network Interface Card) SIP Line container in Interaction
Administrator.
RTP For the hardware platforms, the IP AudioCodes the first RTP session will use port
Address of the IP telephony board.. 4000 (configurable, see section 17.9
“Configuring the AudioCodes Boards with
For software platforms (HMP), the IP Interaction Administrator”). The second RTP
Address of the system’s NIC (Network session will start at an even port interval
Interface Card) number 10 higher than 4000. For example, if
the starting port was 4000, then the first IP
resource will consume 4000 (for RTP), 4001
(for RTCP) and 4002 (for T.38 fax). The next

IP resource will consume 4010, 4011, 4012
and so on. If this was a 120 port card, the
last IP resource will consume 5190 (for RTP),
5191 (for RTCP) and 5192 (for T.38 fax).
Synchronous RTP is used.
For HMP, the first RTP session will use port
49152 (configurable, see section 18.6
“Configuring RTP Dynamic Port Range”). The
second RTP session will use a port number
two high then then the first (the first RTP
session will use port 49152, the second will
use 49154,…). Synchronous RTP is used.
RTCP For the hardware platforms, the IP This will always be one higher than the port
Address of the IP telephony board. number used for its RTP session.
For software platforms (HMP), the IP
Address of the system’s NIC (Network
Interface Card)
Interaction IP Address of the Interaction Center 2633, registered with IANA

Center Notifier system (http://www.iana.org/assignments/port-
process numbers ).
Interaction IP Address of the Interaction Center 3508, registered with IANA

Center Web system (http://www.iana.org/assignments/port-
Services numbers ).
6.3 SIP Q&A

The following questions are some of the most commonly asked questions with regard to SIP and the
Interaction Center.
Q. What differences are there between a SIP version of the Interaction Center and a
TAPI version?
TAPI vs. SIP comparison TAPI Interaction Interaction Center with SIP
Center capabilities
Can the Interaction Center system be No. Yes. SIP was added to the same
mixed with traditional connections via telephony servces component
telephony boards, such as analog that handles all our existing
phones and ISDN trunks? protocols and boards.
What features are lost? A few due to the None.

restrictions of the TAPI
interface that Cisco
provides. See the TAPI
app note for details.
Is a Cisco CallManager needed? Yes. No.
Hardware – do all the gateways, Yes. We have a single No. Multi vendor solutions are
routers, and phones have to be from vendor dependency on used. We have certified phones
Cisco? Cisco. This typically leads from Polycom and gateways from
to higher cost equipment. AudioCodes. Our system can
work with any certified SIP
compliant gateway or phone.
Software – are proprietary or standard Proprietary. Mixture of Standard protocol - SIP.

protocols used to communicate with standard and proprietary
phones and gateways. protocols.

Q. In SIP terms, what are you?
SIP component Does the Interaction Center have the features of

this SIP component?
Application Yes.
Application Server Yes.
Media Server Yes.
User Agent Client Yes.
User Agent Server Yes.
Proxy Yes, the Interaction SIP Proxy is available – or we can

work with any SIP compliant SIP Proxy.
Registrar Yes – or we can work with any SIP compliant SIP

Registrar.
SIP Gateway Yes – because you can add SIP to an existing

Interaction Center with all its working telephony
boards. You can also use the Interaction Center in SIP
only mode, without any analog phone or trunking.
Q. What features do you lose using a SIP IP phone compared to an analog phone?
A. None. In fact, you gain features by using a SIP phone. SIP phones can send SIP compliant
messages hold, transfer, and conference calls. The following table shows some of the
advantages of a SIP phone over a traditional analog phone.
Analog Phone SIP Phone
Must be a hard phone. Can be either a hard or soft phone.
Takes up a dedicated “station” Uses a resource only when in use.

resource (either a station port on a
station board or a T1/E1 channel
when using a channel bank).
Must be locally connected to the Can sit anywhere on the LAN or WAN.
server or to a channel bank.
Flash is used to hold or bring up a Some IP phones have buttons to do hold, transfer,
voice menu to do features like conference,…
conference and transfer.
Q. What features do you lose using a SIP gateway compared to bringing analog or
digital trunks directly into the Interaction Center?
A. None. The Interaction Center server will “talk” SIP to the gateway, and the gateway then
connects to WANs (frame relay,...) or the PSTN (T1, E1, ISDN, Analog). Even features like
recording, call monitoring, call analysis are available over SIP.
Q. What type of SIP phones can be used?

A. Any SIP compliant hard or soft phone that Interactive Intelligence has certified. A reseller
needed more IP phones for testing – so we emailed them a link to free soft IP phones. Many
soft phones are available, even Microsoft Messenger talks SIP. This means any laptop can be a
SIP phone. Many hard phones, from companies like Cisco and Pingtel, have nice features, like
hold, transfer, conferencing, and multiple call appearances. SIP compliant soft or hard phones

can be used as standalone phones or used with the Interaction Client. Also, our Interaction
Client, using Microsoft SIP code (the RTP Client), can act as a SIP phone itself.
Q. Where can these phones physically sit?

A. Phones and gateways can exist anywhere on an IP network. They communicate with the
Interaction Center server via SIP/RTP over IP. This is a very powerful asset but be careful
because VoIP is very sensitive to delays on IP networks and therefore, does require QoS
(Quality of Service) to be configured in your equipment and in your network.
Q. What type of integration do you do with SIP phones? What happens when I hit the
hold button on the phone?
A. The integration is very complete. The second you hit the hold button on the phone, the call
transitions to the held state, the call will show “On Hold” on the Interaction Client, and the
remote user will hear hold music.
Q. What gateways can be used?

A. Any SIP compliant gateway that Interactive Intelligence has certified can be used. This allows
you to have your PSTN connections into the gateway rather than telephony boards in the
Interaction Center.
Q. Describe your connectivity.
A. The Interaction Center can talk directly to SIP phones and SIP gateways, or can send the SIP
request to SIP compliant Proxies, which will do the routing to the SIP phone and gateways.
Q. Is a SIP proxy server required?

A. A SIP proxy server is not required, but does provide some features that might be needed in
certain network topologies. A SIP proxy can do network and also do gateway selection.
Interactive Intelligence offers a SIP proxy server that was designed for remote site use or load-
balancing.

Q. I want the Operator for our company to be able to receive more calls than the
physical IP phone is capable of handling. For instance, I want the Operator to be able
to handle 20 simultaneous calls. Can I do that?
A. Yes, if you want to handle more calls than the IP phone is capable, check the “Persistent”
checkbox in the Station configuration within Interaction Administrator. The Interaction Client
can be used to manipulate a large number of calls while the phone will be the audio device for
the calls. The phone will show one call while the Interaction Client will be used to manipulate
the calls. See section 20 “Defining Global Configurations for SIP Stations” about configuring
Persistent connections.
Q. I want the Call Center Agents for my company to be able to use an IP phone with a
headset, is there any special configuration I need to perform?
A. If a call center agent is using an IP phone with a headset and using the Interaction Client, the
“Persistent” checkbox needs to be selected for the agent’s Station in Interaction Administrator.
See section 20 “Defining Global Configurations for SIP Stations” about configuring Persistent
connections.
6.4 Implementation Overview Diagrams
6.4.1 SIP Hardware Approach Overview

The Interaction Center SIP stack uses SIP to setup and tear down Voice over IP (VoIP) calls. The audio
calls setup with SIP uses the Real Time Protocol (RTP) to transmit voice. Audio from RTP streams gets
put on the internal telephony bus, just like audio from ISDN calls or audio from analog phone sets.
Since all audio from RTP streams is on the telephony bus, all features such as call analysis,
conferencing, recording, monitoring, mixing with analog phones, mixing with trunks (ISDN, E1, T1,
Analog) are available with SIP on the Interaction Center.
SIP can be used for external calls (like ISDN) or to connect to SIP hard or soft phones. SIP phones can
be configured as standalone phones, or used with the normal Interaction Client, or with the
Interaction Center Remote Client. The following diagram shows the architecture of the Interaction
Center with its different telephony options.
Interaction Center Server SIP Soft and

Hard Phones
VoIP Call Control (SIP)
IP LAN
SIP Stack Net wo rk Ca rd
Interaction Center Software
Vo IP Audio (RTP)
IP Cards
Gateway/Routers
Telepho ny Bus
Resources (fax, PSTN

confe rencing, audio) Analog Phones
Tele phony
Code
Analo g Station Cards
SIP Soft and
Hard Phones
ISDN, T1, E1, Ana log PSTN
Trunk Cards
WAN
7 When is a SIP Proxy Needed?

“When is a SIP proxy needed?” is an important question when designing a SIP based Interaction
Center installation. The Interaction Center can sometime alleviate the need for SIP proxies, but in
highly distributed environments a SIP proxy may be needed. Before this question can be answered
it’s first important to understand SIP message routing.

7.1 SIP Message Routing
SIP messages are transmitted over an IP network on a hop by hop basis just like other IP packets.
SIP message routing refers to the process of determining the next hop for a SIP message based on
the SIP message content. The following devices are capable of routing SIP messages,
• SIP phones
• SIP gateways
• SIP proxies
• Interaction Centers
SIP phones only make routing decisions when the user goes off-hook and dials a number. Some
phones with advanced capabilities can route to different destinations based on the dialed number.
Some phones can also detect when the next hop is not responding and route to alternate hops.
Phones with advanced features are typically more expensive than phones without.
SIP gateways mediate between an IP network and the existing PSTN. Gateways receive PSTN calls,
convert them to SIP and then route the SIP messages to the next hop. The next hop is usually
determined by preconfigured routes. Gateways also receive SIP calls and relay them to the PSTN
using one of many PSTN protocols, such as ISDN, T1, etc. Gateways are usually advanced devices
that are capable of routing SIP messages based on dialed number and routing SIP messages so
secondary hops, tertiary hops, etc. Some gateways are even capable of load balancing across a
group of next hops.
Interaction Centers are capable of routing SIP messages between preconfigured SIP stations. A
Interaction Center SIP station is usually a SIP phone. External calls received by the Interaction Center
that are not from a SIP station or not destined for a DID are answered and sent to the auto-attendant.
For outbound calls, the Interaction Center is capable of sending them to different gateways based on
the dialed number or capable of sending them to an outbound proxy.
SIP proxies are designed for SIP message routing. They are usually full-featured and are capable of
many different routing techniques. SIP proxies will be needed with an Interaction Center installation if
the features the customer desires can’t be accomplished by the phones, gateways, and IC servers.
This is discussed in more detail in the sections that follow.
7.2 Phone Specific Routing

Some hard and soft SIP phones can only do simple routing while some can do a considerable amount
of routing.
The following type of routing is typically needed by a SIP phone when used with the Interaction
Center. If the phone is not capable of providing that type of routing then a SIP proxy may be needed.
1. If using switchover, the phone must be able to route its SIP messages to either the primary
Interaction Center or, if the primary Interaction Center is not available, to the backup
Interaction Center.
2. If WAN redundancy at remote sites is required, a phone must be able to route its SIP messages
to either the primary Interaction Center, or the backup Interaction Center if switchover is used,
or to a local gateway if the WAN is not available. This requires the phone be capable of
routing to three destinations.
3. If local or emergency (911) dialing at remote sites is required, a phone must be able to route
its SIP messages to either the primary Interaction Center, or the backup Interaction Center if
switchover is used, or to a local gateway for local or emergency dialing. This requires the
phone be capable of making routing decisions based on the dialed number.
7.2.1 When is a SIP proxy needed for the SIP phones I’m using?

See the SIP 3rd Party Component Feature Matrix spreadsheet for the values in the decision tree below.
These guidelines do not apply if dialing from the IC Client; they only apply if dialing from the phone.
Q. The network has both a primary and a backup Interaction Center. Is a SIP proxy
required?
A. A proxy may be required, depending on the capabilities of the SIP phones. Check the phone’s
“Backup Proxy” capability in the SIP 3rd Party Component Feature Matrix spreadsheet. If
“Backup Proxy” is “Yes” then the phone doesn’t require a SIP proxy. If “Backup Proxy” is “No”,
then a SIP proxy will be needed if the user is dialing from the phone. Note: If the user is
dialing from the Interaction Client, no SIP proxy is needed. Why? Because when the IC
Client makes a call, it sends a proprietary request to the Interaction Center server, which will
place a call to the phone. The phone does not make any routing decisions.
Q. I have a phone in a remote site with a remote gateway. I want emergency calls
and/or local calls to go immediately out the gateway. Is a SIP proxy required?
“Dial Plan Routing” capability in the SIP 3rd Party Component Feature Matrix spreadsheet. If
“Dial Plan Routing” is “Yes”, then no proxy is needed to do this routing. If “Dial Plan Routing”
is “No”, then a SIP proxy will be needed if the user is dialing from the phone. Note: If the
user is dialing from the Interaction Client, no SIP proxy is needed, but the Interaction Center’s
dial plan needs to be configured to route the calls to the local gateway.
Q. The network has both a primary Interaction Center and a local gateway to be used
when the primary Interaction Center is unreachable (no backup Interaction Center is
used)? Is a SIP proxy required?
“Backup Proxy” capability in the SIP 3rd Party Component Feature Matrix spreadsheet. If
“Backup Proxy” is “Yes” then the phone doesn’t require a SIP proxy. If “Backup Proxy” is “No”,
then a SIP proxy will be needed if the user is dialing from the phone.
7.2.2 Gateway Specific Routing

SIP Gateways offer different routing capabilities. The more routing capabilities the gateway has, the
less chance a proxy is required. However, the more gateways are used in the network topology, the
more a proxy becomes a convenient, central location for configuration and for load balancing.
For example, a Cisco gateway can route calls to multiple destinations:
• To a primary Interaction Center, proxy, or gateway (via normal configuration)
• To a backup Interaction Center, proxy, or gateway (via normal configuration)
• To a bank of Interaction Centers (load balancing)
7.2.3 When is a SIP proxy needed for the gateways I’m using?
The same rules apply for the gateways as for the phones.
8 Telephony Connectivity Options with the Interaction Center

One or more of the following telephony connectivity options are available in a single Interaction Center
system.

8.1 PSTN Connectivity Options
1. No gateway (traditional connections, such as ISDN, from carrier). See [1] below.
2. Traditional gateways (ISDN, T1, E1, Analog). See [2] below.
3. SIP gateways. See [3] below.
4. No gateway (IP direct from carrier). See [4] below.
Interaction Center
ISDN, T1, E1, Analog

1 PSTN
2 PSTN / WAN ISDN, T1, E1, Analog
SIP
PSTN / WAN LAN
3
SIP
4 PSTN / WAN LAN

IC Servers with IC Servers with IC servers with IC Servers with
no gateways, ISDN SIP connections no gateway,
using ISDN connections to to gateways using SIP
connections to gateways connections to
the PSTN PSTN/WAN
Gateway Features No gateway. PSTN Connect to the Connect to the No gateways

connectivity is PSTN and WAN via PSTN and WAN via necessary. PSTN
done via the tradition traditional and WAN
telephony boards. connections connections connectivity is
(ISDN, Frame (ISDN, Frame done via SIP. This
Relay) and then Relay) and then is not available
connect to the IC convert all traffic yet, but is coming
server via to SIP. soon by large
traditional carriers.
connections
(ISDN,…).
Are Telephony Tradition ISDN (or Tradition ISDN (or Optional. With Optional. With
boards needed? T1, E1, Analog) T1, E1, Analog) the hardware the hardware
telephony boards telephony boards platform platform
are used to are used to (telephony (telephony
connect to the connect to the boards), IP boards boards), IP boards
PSTN. gateway. are used to do the are used to do the
do the RTP and do the RTP and
transcoding. transcoding.
With the software With the software
platform (Intel platform (Intel
HMP), HMP),
For switchover Yes. The Yes. The No. All No. All
(primary and traditional traditional connections to the connections to the
backup IC connections (such connections (such IC server are done IC server are done
servers), is a data as ISDN) go as ISDN) go via SIP. With SIP, via SIP. With SIP,
probe needed to through the data through the data the switchover the switchover
route the digital probe, which probe, which routing is done routing is done
lines? routes the routes the over the LAN. over the LAN.
connections to the connections to the
appropriate appropriate
server. server.
N+1 Configuration The calls are The calls are The calls are The calls are
(multiple IC distributed, by the distributed, by the distributed, by the distributed, by the
servers) PSTN, across the gateways, across gateways, across PSTN, across the
IC servers, by the IC servers, by the IC servers, IC servers, simply
sending the call to sending the call to simply by sending by sending the SIP
different ISDN different ISDN the SIP messages messages to
trunks. trunks. to different IP different IP
addresses. addresses.

8.2 Phone Options
1. Analog Phones. See [1] below.
2. SIP Phones. See [2] below.
3. Media Gateways. See [3] below.
Analog Phones or SIP Co mpliant Soft

PBX Digital
Phone Media Gateway
Phones with or
Phones without Interaction
IP LAN
Client
3 SIP Co mpliant Hard
Phones with or
without Interaction
2 Client
Interaction Client
Interaction Center used for Audio
1
2 SIP Co mpliant Soft
Analog Phones
Phones with or
without Interaction
IP WAN
Client
SIP Co mpliant Hard

Phones with or
3 without Interaction
Client
Analog Phones or Phone Media Gateway
PBX Digital Interaction Client
Phones used for Audio
1 2 3
IP Phones SIP Phone Media Gateways Analog Phones
Is SIP used to Yes. Yes. The IC server communicates No. Tradition T1/E1
communicate to the with the Phone Media Gateway with boards for channel banks,
phones SIP. The Phone Media Gateway or analog station boards
then communicates with the phone are used to connect to
the same way a traditional channel analog stations.
bank does.
Are resources used No. IP resources are No. IP resources are only used when Yes. The phone uses a
when phone is idle? only used when there is a voice connection. physical resource even
there is a voice when it is idle.
connection.
Does the phone No. The SIP hard or No. The Phone Media Gateway is Yes. The phone has a
have to be directly SIP soft phones are simply an IP device anywhere on the physical connection to the
connected to IC simply IP devices network (LAN or WAN). IC server.
server? anywhere on the
network (LAN or
WAN).
Phone Types Many vendors make Standard analog phones (2500 sets) Standard analog phones
supported SIP hard and SIP and PBX digital phones can be (2500 sets).
soft phones. connected to a wide variety of Phone
Media Gateways.

8.3 Remote Survivability and Emergency Dialing
SIP makes remote survivability straightforward. Calls originating from the phones at a remote site
can be sent directly (via the phone’s dial plan or via a SIP proxy or the IC dial plan) to the remote
gateway for emergency dialing (911), for local dialing, or if the central site is not reachable (remote
survivability). The phone generates its own dial tone, and then based on a variety of configurable
features, such as number dialed or the ability to reach the central site, the call can be sent directly to
a local gateway rather than to the central site.
First, let’s understand Cisco’s two approaches to SRST (Survivable Remote Site Telephony):
Proprietary/CallManager and the SIP approach. Both methods are very similar, the main difference is
that one is a standard and one is proprietary.
8.3.1 Cisco’s NON-SIP SRST (Survivable Remote Site Telephony)
Proprietary SRST Overview
Central Site Remote Site

with Cisco WAN
CallManagers LAN
PSTN NON-SIP SRST capable router with limited set of CallManager

features.
Not shown: Every remote site requires backup central site connectivity.
Outbound: The phone at the remote site can not reach the Cisco CallManagers at the central
site. It will then send the outbound call request to SRST capable router running at its remote
site. The SRST capable router will route the call according to its configuration, typically
using the router’s own connection to the PSTN.
Inbound: An inbound call is received by the a gateway at the remote site and the gateway can
not reach the Cisco CallManagers at the central site. It will then send the call to a SRST
capable router running at its remote site. The SRST capable router will route the call
according to its configuration, typically to a phone at the remote site.

8.3.2 Cisco’s SIP SRST (Survivable Remote Site Telephony)
The exact same logic can be done with SIP and no call managers. The gateways, proxies, and
phones can be easily configured to do alternate routes.
Cisco’s SIP SRST Overview
Central Site
with Remote Site
Interactive WAN
Intelligence’s LAN
Interaction
Centers
PSTN
SIP capable SRST
SIP Proxy (optional)
Cisco Router
Outbound: The phone at the remote site can not reach the Interaction Center Server at the
central site. It will then send the outbound call request to SRST capable router running at its
remote site. The SRST capable router will route the call according to its configuration,
typically using the router’s own connection to the PSTN.
Inbound: An inbound call is received by the a gateway at the remote site and the gateway can
not reach the Interaction Center Server at the central site. The gateway (a SRST capable
router) will route the call according to its configuration, typically to a phone at the remote site.

8.3.3 Interactive Intelligence’s Remote Survivability using SIP
Again, using SIP provides the flexibility of equipment and vendors. Even Cisco’s routers support SIP.
Standard SIP Approach for Remote Survivability

Central Site
with Remote Site
Interactive
Intelligence’s WAN
Interaction LAN
Centers
PSTN
SIP Router SIP Proxy (optional)
Outbound: The phone at the remote site can not reach the Interaction Centers at the central
site. It will then send the SIP outbound call request to a SIP capable router running at its
remote site. The SIP capable router will route the call according to its configuration,
typically using the router’s own connection to the PSTN. Note that if the phone is not
capable of making routing decisions based on unreachable systems, then either a router (which
could do all the SIP routing decisions with its dial plan) or a SIP proxy is needed at the remote
site.
Inbound: An inbound call is received by the gateway at the remote site and the gateway can
not reach the Interaction Centers at the central site. It will then send the call to a SIP capable
router running at its remote site. The SIP capable router will route the call according to its
configuration, typically to a phone at the remote site. Note that if the router is not capable of
routing decisions based off of unreachable systems, then a SIP proxy is needed at the remote
site.
8.3.4 Emergency (911) dialing using SIP
Standard SIP Approach for 911
Central Site
with Remote Site
Interactive
Intelligence’s WAN
Interaction LAN
Centers
PSTN
SIP Router SIP Proxy (optional)
The phone at the remote site dials 911. It will then send the SIP outbound call request to a SIP
capable router running at its remote site, rather than to the Interaction Centers at Central
Site. The SIP capable router will route the call directly to the PSTN. Note that if the phone is
not capable of making routing decisions based on unreachable systems, then either a router
(which could do all the SIP routing decisions with its dial plan) or a SIP proxy is needed at the
remote site.

9 Audio Path and Resource Usage
9.1 Dynamic Audio

The Interaction Center interfaces with SIP messages to the end points (phones, gateways,…).
However, while the SIP messages will always go through the Interaction Center server, the audio can
(or can not) go through the IC server.
The audio will go to the IC server if:

• advance features are needed (and these advance features are not available on the Interaction
Media Server). The advanced features are IVR, attendant, supervisory record, supervisory
monitor, music on hold, and conferencing. When an advance feature is ‘active’, the audio will
be routed through the IC server or the Interaction Media Server. When the advance feature
is finished, the audio will be routed directly between the endpoints.
When the audio is routed directly between the endpoints, no resources (besides a extremely small
amount of CPU and an extremely small amount of network traffic) are used by the IC server for the
SIP messages.
9.1.1 Remote Sites Without Remote Gateways

All calls that originate from the remote phones are sent to the Interaction Center at the Central Site.
The advantage of this is that every call can be recorded and monitored, calling can be done from the
Interaction Client or the phone, and every call shows on the Interaction Client. The disadvantage of
this is the bandwidth required across the WAN link for the remote call to reach the Interaction Center
(if an Interaction Media Server is not available at the remote site)
9.1.2 Remote Sites with Remote Gateways

In release 2.2 of the Interaction Center, the audio flows from the phone at the remote site to the
Interaction Center then from the Interaction Center to the gateway. If the gateway is at the central
site, then there aren’t any problems. If the gateway is a telephony card in the Interaction Center
server, then there aren’t any problems. If the gateway is at a remote site, the audio will be taking
two trips across the WAN, which will use bandwidth and add delay.
Options if the gateway is at the remote site AND that gateway is to be used for inbound and outbound
dialing:
1. IC 2.2 will have the audio take two trips across the network, one from the phone to the
Interaction Center at the remote site, and the second from the Interaction Center to the
remote gateway. All of the features of the Interaction Center are immediately available but the
traverse the WAN connection twice and require 2 calls worth of bandwidth.
2. Starting with IC 2.3, IC will redirect the audio so the audio stays at the remote site (the audio
is not sent to the central site). If an a audio feature is needed and an Interactiom Media
Server at the remote site can not do that feature, then the audio will be sent to the IC server.
3. Some calls originated from the remote phones can be sent directly (via the phones’ dial plan or
a remote proxy) to the remote gateway for emergency dialing (911), for local dialing, or if the
central site is not reachable. In this case, the calls audio does not take a round trip to the
central site, but the Interaction Client cannot be used for this type of dialing (the dialing must
be done from the phone). Also, the central site Interaction Center server is not aware that the
call was made (no recording or no monitoring capabilities, call does not show on the Interaction
Clients).
9.2 Conference Resources
Conference button on phone: There are conference resources on some phone devices, and when the
conference button on the phone is used, the conference mixing is done on the phone itself (no

Interaction Server conference resources are used). The phone will have two separate RTP streams, one
for each party.
9.3 IP Resources
Each audio session (i.e. RTP session) will use an IP resource.
Examples
• A idle IP phone will not use an IP resource.

• An idle SIP gateway will not use an IP resource.
• a call into an ISDN telephony board to an agent using a SIP phone will use one IP resource.
• A call from a SIP gateway to an agent using a SIP phone will use 0 IP resources.
• A call from a SIP gateway to an agent using a SIP phone where an audio feature (recording,
monitoring, conferencing) is needed will use 2 IP resources. If the Interaction Media Server
can do the feature, then 0 IP resources will be used.
9.4 Bandwidth Usage

Each IP session will use 2 half duplex connections. Each connection will use approximately 16 Kbps
for IP header overhead and a additional amount for the voice data: 64Kbps (G.711), 8Kbps (G.729),
6.3Kbps (G.723). So a G.729 session will use 48Kbps (8k for voice, 16k for overhead, and then the
same for the other direction).
A way to reduce the bandwidth usage in half is to use VAD (Voice Activate Detection). VAD will save
bandwidth on silent connections by not transmitting the silence. Since on a normal conversation
there is only one talker and one listener, using VAD will cut the bandwidth roughly in half. So, a
G.729 session using VAD will use 24Kbps (24Kbps for the talker and 0k for the listener because of
VAD). VAD should not be used when bandwidth is plentiful because it can cause choppy speech and
loss of a few syllables at the beginning of audio.
9.5 Sample Systems

See section 16 “Platforms” for all the hardware options. Here are a couple sample, all SIP systems.
Sample 1: 60 agent call center, 2 to1 call ratio (60 active calls connected to agents, 60 calls waiting in
queue), conferencing, faxing.
• Need 180 IP resources (120 IP resources for external calls from the SIP Gateway, 60 IP
resources for the phones). This allows 60 callers to be connected to agents, and 60 callers to
be listening to audio while in agent-wait-state.
• Need voice resources audio (IVR, music on hold, audio in queue)

• Need conference resources
• Need fax resources for incoming faxes
The configuration would take two AudioCodes board (IP resources) and one Aculab board (voice,
conferencing, and fax resources).
Sample 2: 480 business users using 480 SIP stations (i.e. managed phones) and in the worst case, 1
our of 4 phones will be in used at any given time. Therefore, the 480 SIP phones will only use up to
120 IP resources at any given time.
• Need 120 IP resources.
• Need voice resources audio (IVR, music on hold)
• Need conference resources

• Need fax resources for incoming faxes
The configuration would take one AudioCodes board (IP resources) and one Aculab board (voice,
conferencing, and fax resources).
9.6 Putback Transfers (similar to ISDN RLT “Release Link Transfers”)

Putback Transfers are very similar to Dynamic audio, with one important distinction. When a putback
transfer is completed, the Interaction Center relinquishes all call control (the call will not show on the
Interaction client, no operations can be performed all the call). When Dynamic audio is used, the
SIP messages are still going to the Interaction Center, so the ability to control the call is still available
(the call will show on the Interaction Client, operations can be performed on the call).
9.7 For complete details, see section 19.11 “Line Configuration: Access Page
Line Configuration
Granted Access | Denied Address Granted Access: By default, all IP addresses will be allowed
access to the IC server except those listed in the list below.
Denied Access: By default, all IP addresses will be denied
Access | IP Address Put the IP addresses in the list. It’s possible to enter a single IP
address or a range of IP address.

9.8 Line Configuration: Region Page
See the “IC Regionalization and Dial Plan” tech note for details of regionalization.
Line Configuration: Call Putback Page”.
9.9 External Audio Path
9.9.1 External Audio Path Resource Usage
Resource Usage AudioCodes Boards HMP

Table
SIP RTP SIP RTP
Host CPU Used Host CPU used 0 – no RTP Host CPU used 0 – no RTP
(external audio) for SIP comes to IC for SIP comes to IC
messaging server messaging server
processing processing
Host CPU Used Host CPU used 0 – all RTP Host CPU used Host CPU used
(internal audio) for SIP processing is for SIP for RTP
messaging done on boards messaging
processing processing
# IP resources used 0 – SIP stack 0 – no RTP 0 – SIP stack 0 – no RTP

(external audio) does not use IP comes to IC does not use IP comes to IC
resources server resources server
# IP resources used 0 – SIP stack 1 for every call 0 – SIP stack 1 for every call
(internal audio) does not use IP leg does not use IP leg
resources resources
Devices
• External Device A (IP phone, IP gateway,…)
• Interaction Center

• External Device B (IP phone, IP gateway,…)
Scenario
• Inbound call from A to Interaction Center (IVR, dial by name, fax detection …).
• Call transferred to Device B
Configuration
• Both A and B are configured in IA as with an AudioPath of Dynamic.

• A and B could have codecs configure or configured to determine their own codecs with the
AudioPath is dynamic.
9.9.2 External Audio Codec Selection (Device A to IC)
Direction AudioPath SIP Message Details
A to IC Internal or INVITE Contains A’s advertised codecs.

External
IC to A Internal or OK IC will use the find the first codec in the codec list configured
External for Device A in IA that matches a codec in A’s advertised
codecs and return that codec in the OK.
A to IC Internal or ACK Audio can now start.

External
9.9.3 External Audio Codec Selection (IC to Device B)
Once the call’s destination is discovered (ACD agent becomes available, extension dialed, user’s name
dialed,…), IC will send the call to Device B.
IC to A or Internal INVITE Session between A and IC already sent up.

A to IC
IC to B Internal INVITE The INVITE contains the codec list configured for
Device A in IA and IC’s IP address and port number.
IC to B External INVITE Contains either:

First, find which of A’s codecs are going to be used in a
calculation. The intersection of A’s advertised codecs
and A’s configured codecs will be used. We will call
this codec list L.
If normal media, send the intersection of L and B’s
configured codecs.
If delayed media, send the intersection of L, what B
advertised, and B’s configured codecs.
A’s IP address and port number will also be sent.
B to IC Internal or OK The OK contains B’s advertised codecs.

External
IC to B Internal or ACK Audio can now start for Internal.

External

IC to A External Re-INVITE First, find which of B’s codecs are going to be used in a
calculation. The intersection of B’s advertised codecs
and B’s configured codecs will be used. We will call
this codec list L.
Send the intersection of L and A’s configured codecs.
B’s IP address and port number will also be sent.
Rules
• Devices must be SIP (i.e a SIP gateway and an IP phone; or two IP phones). An ISDN trunk
coming into the IC server will always be internal audio.
• To insure G.729 is used by remote phone, you must make that the only codec configured.
Otherwise, another codec could be used.
10 Voice Issues on Networks

IP transmissions are broken into packets that can travel different routes and arrive at different times.
Voice quality is affected by lost packets, delayed packets, and the delay variation of the packets. If the
packet delay (latency) is too great (over 250ms), then the conversation starts to sound bad (like a
cheap pair of walkie-talkies). If the delay is variable, then jitter buffer overruns can occur. If packet
loss occurs, then voice clipping and slips occur.
10.1 Quality of Service (QoS)

Quality Of Service (QoS) refers to the mechanisms in the network that make the actual determination
of which packets have priority. Cisco has put a lot of effort into QoS in their router and gateway
operating systems.
Note: Networks must be capable of the extra burden of voice traffic bandwidth. Networks must be
designed and configured for QoS (i.e. have a voice infrastructure in place), or the voice quality will not
be acceptable. The Interaction Center will send voice packets on to the network and will set the
associated QoS fields accordingly. Interactive Intelligence will not be able to fix voice quality problems
you encounter since they are usually caused by the network configuration or design. Available on the
Cisco website are two good resources for understanding the importance of QoS. They are QoS
overview and QoS design guide.
10.1.1 Layer 3 IP Header Byte

Each voice or data datagram has a byte in the IP header that provides the quality of service treatment
desired. Routers, switches, and gateways must be configured to observe this byte. This byte,
depending on the protocol, can be used differently.
Interactive Intelligence vendor specific settings of the Type of Service byte:
1. AudioCodes IP boards set this byte, by default, to 0xA0 (=1010 0000), and can be changed in
the line configuration section in Interaction Administrator.
2. Intel HMP set this byte, by default, to 0x00, and can be changed in the line configuration
section in Interaction Administrator.
3. Cisco IP phones set this byte, by default, to 0xA0 (=1010 0000).
4. Messenger 4.6 sets this byte to 0x00.
5. Messenger 4.7.0041 sets this byte to 0xA0 (=1010 0000).
10.1.1.1 Layer 3 IP Header Byte (ToS)

Here, the byte is described according to RFC 791 “Internet Protocol”.

Layer 3 Ipv4 Type of Service Byte (RFC 791, section 3.2)
Bit 0 Bit 1 Bit 2 Bit 3 Bit 4 Bit 5 Bit 6 Bit 7
IP Precedence Type Of Service (RFC 1349)
111: Network Control – intended to Delay Throughput Reliability Cost Unused

be used within a network only.
0: Normal 0: Normal 0: Normal 0: Normal 0
110: Internetwork Control –
intended to be used by gateway 1: Low 1: High 1: High 1: Low
control originators only
101: CRITIC/ECP
100: Flash Override
011: Flash
010: Immediate
001: Priority
000: Routine
10.1.1.2 Layer 3 IP Header Byte (DiffServ)

Here, the same byte is described according to RFC 2474 “Definition of the Differentiated Services
Field (DS Field) in the IPv4 and IPv6 Headers”.
Layer 3 Ipv4 Differentiated Services (RFC 2474, section 3)
Bit 0 Bit 1 Bit 2 Bit 3 Bit 4 Bit 5 Bit 6 Bit 7
DSCP (Differentiated Services Control Protocol – DiffServ) DSCP Flow Control
10.1.2 Layer 2 Byte (802.1p/Q)

All telephony IP boards (in the IC servers) and IP Phones should be in the same VLAN (Virtual LAN).
Voice traffic needs to be isolated from general network traffic. A VLAN (Virtual Switch might be a
better name) can be created from a group of ports on a switch (and can span multiple switches). For
instance, ports 3 and 4 on Switch 1 and port 5 from Switch 2 can be included in a VLAN. If all the
telephony IP boards and IP Phones were on these ports, voice traffic would be isolated from general
network broadcasts and quality can be improved.
For QoS, Layer 2 802.1p/Q signaling method sets priority levels in both RTP and RTCP packets (in the
VLAN tag field). Telephony IP boards do NOT set this layer 2 byte but rely on the switch to do this.
10.1.3 Notes about QoS and the Interaction Client.
Also see section 18.5.2 “QoS Setting”.

From the client's point of view, QoS is a network issue, not a programming one.
There are three layers used by the client for SIP processing: The Client layer, the RTC layer, and the
Generic QoS (GQoS) layer.
Client layer
The client does not have the ability to set DSCP bits directly. This work is done at a lower level. The
Client communicates directly with the RTC layer.

RTC layer
The RTC layer defines specific flow-specs for audio and video, internally, and through PSched the
DSCP (TOS is deprecated) will happen. RTC uses lot of parameters like network delay, latency,
packetloss, link speed, etc to determine the best possible configuration to use for streaming
audio/video. The RTC layer communicates directly with the GQoS layer, and requests guaranteed QoS
for audio flows. If the GQoS system is not in place on the network, the bits will have their default
value, 0 (no priority).
GQoS layer
The GQoS layer controls Quality of Service. This layer will set the DiffServ bits, but only if the network
has the whole Microsoft QoS system in place. Normally, the values are policy defined, and the defaults
can be overridden from gpedit.msc. Launch gpedit.msc and go to: "Local Computer
Policy\Administrative Templates\Network\QoS Packet Scheduler\DSCP ..." QoS aware applications can
request a certain service type for a traffic flow from the GQoS layer. Available services are:
• Guaranteed service — offers high quality, quantifiable guarantees with bounded latency.
• Controlled load service — offers high quality, quantifiable guarantees to approximate the
service that would be provided by a lightly loaded network.
• Qualitative service — generally offers medium quality, non-quantifiable guarantees
10.2 Echo
Cisco’s whitepaper Echo Analysis for Voice Over IP gives a very good overview for solving echo issues
in pure VoIP or hybrid VoIP environments.
10.3 RTCP Sender Reports

The Interaction Center logs the cumulative data from RTCP sender reports (SR) that are transmitted
and received during an RTP audio session. The reports provide quality feedback from both the senders
perspective. The IC will always log a local report which is an accumulation of quality feedback data
that originates from the IC’s internal IP boards. If the remote site transmits SR’s then the IC will log a
remote report which is an accumulation of quality feedback data from the remotes perspective. For
details of the RTCP sender report see RFC1889.
The local and remote reports are logged for every SIP call in the TsServer.Vwrlog under the RTPQos
topic. The log file is located in the \I3\IC\Logs directory. These reports are logged with the default
trace settings (no tracing has to be enabled for these two lines to be logged).
RTPQoS:IPLinkResource.cpp(571):CIPResourceMgr::LogAudioCodesQoSData(): 1100092694
ACB0C18 local report: tx packets 19105, tx octets 3056640, rx lost packets 37,
jitter (hi) 36, jitter (lo) 0, jitter (avg) 16
RTPQoS:IPLinkResource.cpp(584):CIPResourceMgr::LogAudioCodesQoSData(): 1100092694
ACB0C18 remote report: tx packets 17293, tx octets 2766880, rx lost packets 0,
jitter (hi) 16, jitter (lo) 0, jitter (avg) 10
The data in the report consists of the following fields:
Data Definition
Local Report Data generated by the Interaction Center server.
Remote Report Data generated by the remote end point.
tx packets The total number of RTP data packets transmitted by the sender since starting
transmission.
tx octets The total number of payload octets (i.e., not including header or padding) transmitted
in RTP data packets by the sender since starting transmission

rx lost packets The total number of RTP data packets from source that have been lost since the
beginning of reception. This number is defined to be the number of packets expected
less the number of packets actually received, where the number of packets received
includes any which are late or duplicates. Thus packets that arrive late are not
counted as lost, and the loss may be negative if there are duplicates.
jitter (avg) An estimate (in milliseconds) of the statistical variance of the RTP data packet inter-
arrival time, measured in timestamp units and expressed as an unsigned integer. The
inter-arrival jitter J is defined to be the mean deviation (smoothed absolute value) of
the difference D in packet spacing at the receiver compared to the sender for a pair of
packets.
N/A implies this data is not available (not available with Intel/Dialogic products).
jitter (hi) Maximum jitter (in milliseconds) recorded over session.

jitter (lo) Minimum jitter (in milliseconds) recorded over session.

The inter-arrival jitter field provides a measure of network congestion. Packets lost tracks persistent
congestion while the jitter measure tracks transient congestion. The jitter measure may indicate
congestion before it leads to packet loss. Since the interarrival jitter field is only a snapshot of the
jitter at the time of a report, it may be necessary to analyze a number of reports within a single
network.
The packets and octets count provide a good indication of the network bandwidth requirements for the
session. They can be used to determine if the network is properly sized for the amount of voice traffic
it receives.
11 Security
Interactive Intelligence recommends using SIP access over a WAN by utilizing a VPN; opening port
5060 (the default port used for SIP) in corporate firewalls is NOT recommended since that port will
doubtless become a target of hackers as SIP becomes more ubiquitous.
Line side and station side authentication is supported. See the authentication configuration
descriptions in the line configuration (section 19.7, “Line Configuration: Authentication Page”) and in
the station configuration (section 21.7 ”Station Configuration: Authentication Page”).
11.1.1 Security Alert

Currently, there are security alerts for the 2 most common VoIP protocols, H.323 (which is not used
by Interactive Intelligence) and SIP (which is used by Interactive Intelligence).
Several critical flaws have been discovered in VoIP products based on the widely used H.323 protocol.
The flaws are outlined in CERT® Advisory CA-2004-01 Multiple H.323 Message Vulnerabilities. Note that
Interactive Intelligence does not use H.323, it has chosen SIP exclusively as its VoIP protocol.
Flaws have also been discovered in VoIP products based on the emerging SIP protocol. The flaws are
outlined in CERT Advisory CA-2003-06 Multiple vulnerabilities in implementations of the Session Initiation Protocol
®
(SIP). Interactive Intelligence continues to test its SIP product lines for against unauthorized
privileged access and denial of service attacks.
11.1.2 IC features
Unauthorized Access: We have added authentication as specified in RFC 2617. Interactive

Intelligence products can pass encoded user names and passwords for usage access. Work has almost
completed in receiving authentication from stations using this same mechanism.

Denial of Service: This will become more and more important as customers advertise public SIP
addresses (call 800-555-1212 or sip.inin.com). We plan to test our stack with the OUSPG test suite
against malformed SIP messages that can cause any undesired behavior. Flooding detection can be
done by our system - or by a SIP proxy. Since our initial SIP release of the 2.2 Interaction Center in
June of 2002, we have configuration limiting the number of inbound calls a system will allow - but this
does not address the problem of using all these resources by an attacker sending multiple, legitimate
SIP inbound requests. This would be similar to a system using all its inbound ISDN trunks to a set of
attackers. Planned is even more detection logic and throttle logic to address this situation.
Access Control Lists: In 2.4, ACLs are configurable in the SIP line configuration.
12 Firewalls and NAT

Firewalls and Network Address Translation (NAT) WILL block IP voice and video. There are basically 4
ways to address this problem (see below). How you do this might be determined whether you control
the equipment that is providing the firewall and NAT.
Solutions
1. RECOMMENDED: Tunnelling. Tunnel through the firewall and NATs by using VPN to tunnel.
This is the solution recommended by Interactive Intelligence. Opening port 5060 (the default
port used for SIP) on corporate firewalls is NOT recommended since that port will doubtless
become a target hackers as SIP becomes more ubiquitous
2. Bypass. Bypass the firewall and NATs by moving the equipment outside the firewall.
3. SIP enabled firewalls. Upgrade your firewalls to understand SIP or to be able to be controlled
by a SIP proxy. For example, the Cisco Pix 6.1(4) version is SIP-enabled.
4. Transversal. Buy equipment that solves the problem. This is typically a client box on each LAN
and a server box somewhere on your network.
12.1 Cisco Firewall Information

Sample configuration for a Cisco Firewall.
Model: PIX model 520
Software Version: Software version 6.1(4)
Lines needed to enable SIP
fixup protocol sip 5060
Lines needed to NAT IP addresses
static (inside,outside) <public ip address of IC server> <private ip address of IC
server> netmask 255.255.255.255
static (inside,outside) <public ip address of IP card> <private ip address of IP
card> netmask 255.255.255.255
Lines needed to open the SIP port
conduit permit udp host <public ip of IC server> eq 5060 any

or
access-list <id> permit udp any host <public ip of IC server> eq 5060
12.2 VPN
Setting up a station on the WAN connected over VPN is identical to configuring a station on the LAN.
Note that when a remote station VPNs into the network, it is given a local IP address. Since this IP
address can change on every instance of connecting over VPN, the contact address in the station

configuration should be the “name” of the station, rather than the station’s IP address. The station
contact information is configured in section 21 “Creating and Configuring SIP stations in Interaction
Administrator”.
13 Notes About User and Station Extensions

With the Interaction Center, you have the ability to configure extensions for users and extensions for
IP phones. Depending on configuration, the call will take:
• user logic: If the user is not in an available status, the call will roll to voice mail. If the user is
in an available status, the call will go first go to the phone(s) that the user is logged into (i.e.
follow the user). If the user is not logged in, the call will go to the user’s default workstation
(set in the user configuration in Interaction Administrator). If there is not a default workstation,
the call will roll to voicemail.
• or station logic: The call will ring the station, no voice mail.
User and station extensions must unique extensions (i.e. user extensions are different than phone
extensions). This allows users to “roam”, which means a user can be associated with any phone (by
logging in or starting a client) and his calls will follow him.
14 Inbound Logic
14.1 Diversion
A diversion header in a SIP message looks similar to this:
Diversion: <sip:5858652@siptest.wcom.com>;reason=no-answer
Notes
• If the CC_Diversion header is received, the Interaction Center treats it as equivalent to the
Diversion header
• If multiple diversion headers are received (or multiple entries in a single diversion header), the
top most header (or first entry) is the last diverted user.
The Interaction Center will set the following values:
Eic_RedirectionTn This is the number that is receiving the redirected call.
Set from the SIP message URI address.
For sip address scheme (addresses that start with “sip:”),
type and port number are added if not present in the header
(sip:user@host:port).
In the above example, Eic_RedirectionTn would be
sip:voicemail@204.180.46.185
Eic_RedirectingTn This is the number that is redirecting the call.

In the above example, Eic_RedirectingTn would be
sip:5858652@siptest.wcom.com
14.2 Inbound Calls, including DID

When a SIP call arrives at the Interaction Center, it has a
• destination address (the SIP URI address) in the form sip:user@host:port
• origination address (the SIP “From” Header) in the form sip:user@host:port

• Optional diversion address (the SIP “Diversion” Header) in the form sip:user@host:port. This
is the address of the device that diverted the call. It also contains the reason the call was
diverted.
The Interaction Center can direct the call to a user, station, or workgroup.
• When a call is directed to a station, it takes station logic (see section 13, “Notes About User
and Station Extensions”).
• When a call is directed to a user, it takes user logic (see section 13, “Notes About User and
Station Extensions”).
The Interaction Center will process the call with the following logic:
1. Check if the call was made from a managed station. If so, make the call on behalf of the
station.
2. Set the following attributes for handlers
Eic_LocalTn Set from the SIP URI (which is sometimes different from the “To”
header address).
For sip address scheme (addresses that start with “sip:”), type and
port number are added if not present in the header
Eic_LocalName Set from the SIP message “To” header display name
Eic_RemoteTn Set from the SIP message “From” header address.

Eic_RemoteName Set to the SIP message “From” header display name
Eic_ContactAddress Set to the SIP message “Contact” header.

This attribute is set if the Diversion header is present.
Example:
INVITE sip:voicemail@204.180.46.185 SIP/2.0
To: <sip:5858652@siptest.wcom.com>
From: "Fred
Flintstone"<sip:cic@i3worldcom.com>;tag=26680
Via: SIP/2.0/UDP
204.180.46.185;received=204.180.46.185
Call-ID:
5df33e3731a72d7309a756b4571016d2@204.180.46.185
CSeq: 1 INVITE
Diversion: <sip:5858652@siptest.wcom.com>;reason=no-
answer
For example, if a phone 5858652 redirects the call to

sip:voicemail@204.180.46.185, Eic_RedirectionTn would be
sip:voicemail@204.180.46.185.
Eic_RedirectingTn This is the number that is redirecting (or diverting the call).
This attribute is set if the Diversion header is present.

See example above for For Eic_RedirectionTn
Diversion:
<sip:5858652@siptest.wcom.com>;reason=no-answer
For example, if a phone 5858652 redirects the call to
sip:voicemail@204.180.46.185, Eic_RedirectingTn would be
sip:5858652@siptest.wcom.com.
Eic_ReasonForCall “U” for Unknown

“B” for Busy
“N” for No answer
“D” for Direct
“A” for Always
In the above example, Eic_ReasonForCall would be “N”
Eic_ReasonForCallString Exact SIP reason in the Diversion header

This attribute is only set if the Diversion header is present.
Eic_OutboundSetupParams Deprecated in 2.3. Use Eic_UserToUserData.
Eic_UserToUserData The value in the call attribute Eic_UserToUserData will be put in the
ININAttr header in the SIP message during outbound call logic and
blind transfer logic.
Eic_UserToUserData syntax:
[name=value[;name=value]*]
Note: name and value can not contain any double quotes.
Note: name must start with uu_.
When a call is received, call attributes name1, name2,… will be set

to value1, value2,…
Example:
If Eic_UserToUserData is set to:
uu_Agent=Frank Smith;uu_Number In Queue=12
Then the outbound SIP message will include this header:
ININAttr: “uu_Agent=Frank Smith;uu_Number In Queue=12”
And the inbound call will have the following attributes set before the
incoming call handler is invoked:
uu_Agent will be set to: Frank Smith
uu_Number In Queue will be set to: 12
Eic_InfoMsgContents Empty or “Addresses”. See section below on the SIP INFO

message.

3. Check Eic_LocalTn’s whole address (sip:user@host:port) to see if it matches a entry in the
configured Interaction Administrator Phone Number DID container (case insensitive match, but
sip: and :port must be present in the entry). If so, the call will be directed to that configured
user, phone, or workgroup. This is done in SystemDNISRouting.ihd (called from
System_IncomingCall.ihd).
4. Check Eic_LocalTn’s user portion of the SIP address (user) to see if it matches an entry in the
configured Interaction Administrator Phone Number DID container (case insensitive match). If
so, the call will be directed to the configured user of phone. This is done in
SystemDNISRouting.ihd (called from System_IncomingCall.ihd).
5. Check Eic_LocalTn’s user portion of the SIP address (user) to see if it matches a user extension
or a station extension. If so, the call will be directed to that user or phone. This is done in
SystemIncomingSIP.ihd (called from System_IncomingCall.ihd). Note that calls to a queue,
such as a ACD queue ( 3@sip.inin.com, where 3 is an ACD queue), will need an entry in the
DID table (Phone number container in Interaction Administrator) for them to be routed
immediately by the system.
6. In SystemIncomingSIP.ihd (called from System_IncomingCall.ihd), check an attribute’s whole
address (sip:user@host:port) or the user portion (user) to see if it is an exact match for special
dialing for the following server parameters:
• Eic_LocalTn is compared against the IP Managed Phone Shortcut server parameter (section
22.4 “Configuring the Managed Phone Shortcut”), or
• Eic_LocalTn is compared against the IP Message Button server parameter (section 22.5
“Configuring Message Waiting Indicators (MWI)”), or
• Eic_RedirectionTn is compared against the IP Voicemail Direct server parameter (section
22.3 “Configuring Voice Mail for Non-Managed Phones”).
7. Do IP VoiceMail Direct logic (see section 22.3 “Configuring Voice Mail for Non-Managed
Phones”). This is done in SystemIncomingSIP.ihd (called from System_IncomingCall.ihd).
8. Check if the user portion (user) is an exact match for special dialing, such as “*” dialing or no
number dialing (encountered when just the “#” is entered). This is done in
System_InitiateCallRequest.ihd.
9. If none of the above match, the call will be treated like a new inbound call and be sent as
configured, probably to a main IVR.
14.3 SIP Info Message

On some analog systems (like a PBX to PIMG analog connection), the caller id is not delivered
until the call is answered, so the data is not available when the Interaction Center receives the
SIP INVITE. A SIP INFO message can be received to set some key attributes.
Eic_InfoMsgContents is set to a constant string “Addresses” to let the handler know the Info
message came in. A handler can wait a certain amount of time if it knows that a SIP INFO
message might be coming.
If a diversion header is received then Eic_ReasonForCall, Eic_ReasonForCallString, and
Eic_ReDirectingTn are reset.
If digits are received in the body of the SIP INFO message, the Eic_RedirectionTn attribute is set.
15 Outbound Logic
When a call is made from using the Interaction Client, an audio path must be made between the SIP
phone and the Interaction Center server. The Interaction Center server will make a call to the SIP

managed station, and then once the connection is made, will complete the requested call. Note that if
persistent connections are used, an audio connection might already be established, which means that
the request call will start immediately.
For connection calls: “From” number logic that is passed in the SIP message is:
• If not persistent, the remote number taken from attributes of the call that cause the
connection call to be made (Eic_RemoteTn or Eic_RemoteAddress)
• Or if the above is empty, the phone number configured in the SIP line
For connection calls: “From” name logic that is passed in the SIP message is:
• If persistent, the name configured in the SIP default display string
• Or if not persistent, the remote name taken from attributes of the call that cause the
connection call to be made (Eic_RemoteName)
• Or if the above is empty, the name configured in the SIP default display string
• Or if the above is empty, the name in the Eic_LocalTn attribute
For regular calls: “From” number field logic that is passed in the SIP message is:
• The number passed in the CompleteExtendCall tool step (in SystemInitiateCallRequest)
• Or if the above is empty, the phone number configured in the SIP line
For regular calls: “From” name field logic that is passed in the SIP message is:
• The name passed in the CompleteExtendCall tool step (in SystemInitiateCallRequest)
• Or if the above is empty, the name in the Eic_LocalTn attribute
Also, when making a call or blind transfer, call attributes can be set on the tools for MakeCall and
BlindTransfer. The following attributes can be used to pass information on the SIP call.
Eic_OutboundSetupParams Deprecated in 2.3. Use Eic_UserToUserData.
Eic_UserToUserData The value in the call attribute Eic_UserToUserData will be put in the
ININAttr header in the SIP message during outbound call logic and blind
transfer logic.
Eic_UserToUserData syntax:
[name=value[;name=value]*]
Note: name and value can not contain any double quotes.
Note: name must start with uu_.
When a call is received, call attributes name1, name2,… will be set to

value1, value2,…
Example:
If Eic_UserToUserData is set to:
uu_Agent=Frank Smith;uu_Number In Queue=12
Then the outbound SIP message will include this header:
ININAttr: “uu_Agent=Frank Smith;uu_Number In Queue=12”
And the inbound call will have the following attributes set before the
incoming call handler is invoked:
uu_Agent will be set to: Frank Smith
uu_Number In Queue will be set to: 12

16 Platforms
Hardware Platform: Two types of IP boards vendors (AudioCodes and Intel/Dialogic) are used to
deliver the audio (RTP sessions) from the network onto the telephony bus. The IP boards are typically
used with other telephony boards (fax, conference, T1/E1/ISDN,…) from Intel or Aculab.
Software Platform: Intel’s HMP (Host Media Processing) is software that complete replaces the
need for telephony board. This is a total software solution.
16.1 Platform Combinations and Supported Status
Telephony IP-enabled Supported?

Platform Telephony
Boards
Intel/Dialogic None Yes. See section 18 for configuration details.

Software needed.
(HMP) Implemented
in software.
Aculab AudioCodes Yes (this is the preferred hardware configuration). See section 17 for
Hardware IP boards AudioCodes IP board configuration details.
Intel/Dialogic No (not in plan). Intel/Dialogic IP boards are not supported.

IP boards
Intel/Dialogic AudioCodes Yes, with caveats. Dialogic plus AudioCodes configuration has been
PCI Hardware IP boards validated for IC 2.2 for the addition of one (1) AudioCodes card in
selected Dialogic configurations. Two or more AudioCodes cards in a
Dialogic system have not been tested and is not supported at this
time.
Please refer to the Validated Server Matrix spreadsheet for existing
installs and for new installs
While no significant issues have been found with the combination of
Dialogic and AudioCodes cards, we are unable to give blanket approval
to older existing servers due to the higher CPU loads required for
AudioCodes SIP processing. There may be older PCI servers in the
customer base that will not perform with AudioCodes cards.
Intel/Dialogic No (not in plan). ISA telephony boards cannot be mixed on same

ISA Hardware system with PCI IP boards (all IP boards are PCI).
16.2 Platform Comparison
AudioCodes IP Boards Intel/Dialogic HMP Software
Hardware 2 flavors of Audiocodes H.100 PCI boards are No telephony boards required. This
supported. Older, retired H.100 Non-Universal is a complete software solution.
PCI Boards and the newer Universal boards
(supported in 2.3 and above). Note that many
servers do not have PCI slots that allow for
Non-Universal boards. Mixture of these boards
in a single server is supported.

H.100 Non-Universal PCI Boards

• IPM260A/120/NoSpans/H100/MVIP (ulaw
or alaw, 120 RTP sessions
• IPM260A/60/NoSpans/H100/MVIP (ulaw or
alaw, 60 RTP sessions).
• IPM260A/30/NoSpans/H100/MVIP (ulaw or
alaw, 30 RTP sessions).
H.100 Universal PCI Boards.
• IPM260A/240/NoSpans/H100/MVIP/UNI/N3
(ulaw or alaw, 240 RTP sessions).
Price Check with your hardware vendor. Check with Interactive Intelligence.
Density 30, 60, and 120 simultaneous RTP sessions. HMP 1.1/1.3: See the HMP chapter
for the latest densities (section 18
Benefit: The actual number of usable “Installing and Configuring Intel
resources ports provided may exceed the rated HMP Software Solution”)
capacity of the Audiocodes boards. Currently,
the Audiocodes 30 port board reports in as a 40
port board. All 40 sessions are usable on the
30 port board, but this is not guaranteed in
future AudioCodes firmware.
Number of New numbers will be coming out shortly with NA. This is a total software
boards per the new worst case scenario numbers. These solution without boards.
Server numbers are what should be used when
installing a system.
Note that 600 ports was tested with a light call
rate (3 calls/second, 180 calls/minute). Also,
all calls were not being recorded and tracing
was set to default.
Aculab Servers: 600 AudioCodes IP ports can
be in a single server (five 120 port boards).
Also, there is an Aculab limitations of 300
simultaneous audio operations (plays and
records) on a single Aculab system. Note,
since the Audiocodes boards are non-universal,
there are only a few servers that can accept
many non-universal boards.
Multiprocessor Yes. HMP 1.1/1.3: HT is only supported

and hyper with P4 or Xeon processors and
threading Windows 2003. With the above,
Capable the following is supported:
• single processor no HT, or
• single processor with HT, or
• dual processors no HT,
• or dual processors with HT
HMP 1.1/1.3 and Windows 2000,

the following is supported:
• Only single processor with no
HT.
HMP 2.0: Yes, quad processor and
processor affinity.
NICs Multiple NICs are supported. HMP 1.1: Dual NICs are not
supported.
HMP 1.3: The one, specific NIC (on
a dual NIC system) that will be
used is configurable via the DCM.
Play and Record No. Must use an additional resource board. I3 Yes.
does not use the AudioCodes voice resources
and the board is priced accordingly. Voice Note: 2.3.1 includes HMP
resources from Aculab or Intel/Dialogic cards transaction record (in earlier
are used for audio. releases without transaction
record, conference resources were
Note: Aculab Prosody boards (each with up to 4 used to do supervisory records).
DSPs and 4 T1s) are supported. Each of the 4
DSPs can be used for either 60 voice resources,
8 faxes, 24 conference resources with echo
cancellation, or 64 conference resources
without echo cancellation.
Note: Intel/Dialogic 240 voice resource board
(DMV2400A-PCI) is a high density voice board.
It can be configured for 240 voice resources, or
120 conference resources, or 60 of each (60 is
not a typo – when configured for both
conferencing and voice, you lose density).
Conferencing No. Must use an additional resource board. I3 Yes.

does not use the AudioCodes conferencing
resources.
Usable T1/E1 The versions of these AudioCodes boards do No. This is a total SIP solution.
Interfaces not have network interfaces. SIP gateways must be used if
T1/E1 interfaces are required.
Telephony Bus H.100/PCI To telephony bus, this is a total

software solution.
Computer Bus NON-universal PCI and Universal PCI (in 2.3). NA
Coders G.711, G.723, G.729, GSM, G.726. HMP 1.3: G.711, G.723, G.729,
G.726.
Note G.726 is new in HMP 1.3.
RFC 2833 DTMF Yes Yes
Echo 30ms of echo cancellation on the voice going HMP 1.1/1.3: No.
Cancellation from the TDM bus to the IP network.
Coming in later release.
4.2 Firmware will increase it to 64 or 128ms.
RFC3389 (VAD Yes No

for G.711)
RTP Starts at 4000 (configurable) and steps up by Starts at 49152 (configurable, see
10 for the next session. section 18.6 “Configuring RTP
Dynamic Port Range”) and steps up
Sync RTP is used. by 2 for the next session.
Sync RTP is used.
Fax Two options: T.38 (available in 2.2 SR-E and

• T.38 (available in 2.2 and beyond) 2.3)
• Clear channel (which is not reliable)
Modems Clear channel (which is not reliable). See Clear channel (which is not
section 27 “Modem Configuration”. reliable). See section 27 “Modem
Configuration”.
Call analysis Yes, with limitations. See section 25 “Call Yes, with limitations. See section
Analysis” 25 “Call Analysis”
Used with Yes No. This is a total software

Aculab solution. No hardware is required.
systems?
Used with Yes No. This is a total software

Intel/Dialogic solution. No hardware is required.
systems?
Vendor IC 2.x uses Audiocodes release 4.2 (with latest IC 2.x uses HMP 1.1
Software hot fix)
IC 2.3.x uses HMP 1.1
IC 2.3.x uses Audiocodes release 4.2
IC 2.4 uses HMP 1.3
IC 2.4 uses Audiocodes release 4.6
Power Usage H.100 [PCI] IPM-260A-120-HIP-CI-x None.

3.0 A (13 watts) @ 5V without the E1/T1
interface
3.6 A (16 watts) @ 5V with the E1/T1 interface
17 Installing and Configuring AudioCodes Boards

Check for support status and vendor combination in section 16.1 ”Platform Combinations and
Supported Status” before using AudioCodes IP Boards.
17.1 Important Notes and Restrictions

AudioCodes boards have made newer versions of their boards. This newer version is a Universal PCI
(compared to the older non-Universal PCI) form factor. The Universal PCI board will fit into the new
servers. The Universal and non-universal boards will be supported in 2.3 and 2.4. 2.2 will only
support the older, non-Universal boards. The non-universal boards will lose their support from
Audiocodes with newer versions of their firmware in the near future.
AudioCodes boards don’t have a special setup program like Dialogic or Aculab. The Interaction Center
install copies all the necessary files as part of the installation process. Since AudioCodes boards will
always be installed with either Dialogic or Aculab boards, just follow the setups for those other boards,
referring to the appropriate Dialogic or Aculab application (Appendix F: Procedure for Configuring
Aculab/AudioCodes System ) notes. Then follow the procedure in the sections below.
17.2 Servers
The Interactive Intelligence we site maintains a list of servers for certified for use with AudioCodes
boards.
17.3 Known Issues

No Issues at this time.

17.4 AudioCodes with Dialogic
17.4.1 Physical Placement of Boards

If placing AudioCodes boards in a Dialogic system, then surround the AudioCodes boards by Dialogic
boards. Why? Dialogic boards will sense if they are the terminating Dialogic board on the H.100
bus, ignoring other vendor’s boards. If the AudioCodes boards are place on the end, then a Dialogic
board next to it might mistakenly sense that it’s a terminating board and software terminate the
H.100 bus.
17.4.2 Dialogic’s 3rd Party Board Configuration

Dialogic’s 3rd Party Device configuration should NOT be done for AudioCodes boards.
17.4.3 Turning off Secondary Clock Master

On some systems (depending on what Dialogic boards are in the system) the secondary clock master
might have to be disabled to get the Dialogic service to start. To turn off the secondary clock master:
• Right click (in the DCM) on “Bus-0”
• Set “Using Secondary Master (User Defined)” to “No”
17.5 AudioCodes with Aculab

If placing AudioCodes boards in an Aculab system, the latest Aculab Application Note indicates how
the cards must be ordered in the chassis for the system to work properly.
17.6 a-law and mu-law

The Dialogic and Aculab Application Notes should be referenced. When not using a line card, the
procedure is more complicated.

17.7 Prerequisites
• Load the Interaction Center software first. The Interaction Center install will copy all the
AudioCodes files that are needed for the AudioCodes configuration.
• Note the MAC addresses of the AudioCodes boards in your system. It is printed on a label on
the board and also on the shipping box. The MAC address will be needed for configuration later
in this chapter.
17.8 AudioCodes PCI Driver

The Interaction Center install program will copy the necessary PCI driver files for the AudioCodes
boards onto the server machine. It will also install them and trigger the Device Manager to recognize
them. AudioCodes’s PCI driver is not a signed driver so you may get a warning from the operating
system stating such. You must allow the driver to be installed otherwise the AudioCodes boards will
not function properly.
If the install fails to install the drivers or you wish to manually install them, try this manual steps:
1. Open a cmd window and change to the IC\I3\Server\Diagnostics\AudioCodes directory.

2. Run the install_windrvr.bat file. This will copy the files to the necessary location and trigger
the Device Manager to begin the registration process.
Follow the following steps to activate the AudioCodes IPM-260 PCI drivers using the Hardware Wizard.
1. Press the Next button when the Found new Hardware Wizard dialog appears.

2. Select Search for a suitable driver for my device [recommended] and then press the Next
button.
3. Clear all of the Optional search locations: and then press the Next button.
4. After a few seconds the wizard should report that it was able to locate a driver for this device
at C:\WINNT\inf\ipm260.inf or c:\WINNT\inf\ipm206_UN.inf for the universal board. Select

Next to select this driver.
5. Select Finish to complete the process.

17.9 Configuring the AudioCodes Boards with Interaction Administrator
After the PCI driver was successfully installed, the next step to using the boards is to configure them
with Interaction Administrator.
IMPORTANT
• If using switchover, configure the boards in BOTH systems on a single system. Switchover
replicates that configuration across both systems. The boards that are configured in IA but are not
found in the server are not activated. So, for example, if you have a single AudioCodes card in
each server then you MUST configure them both in IA. Switchover will replicate that configuration
to the secondary server. When TsServer starts on the primary, it’ll only detect one of the two
boards in the server and only activate it. If a switchover occurs, TsServer will start on the
secondary server, detect only one board (the other one) and activate it.
• The Interaction Center must be restarted for these changes to take place.
Parameter Description
Firmware Path Indicates the location of the IPM-260 firmware file. Typically, this should be
“D:\I3\IC\Server\Firmware\AudioCodes\ramIPM-260.hex”. It should
contain the complete path, including the firmware file name.
H.100 Bus Law Type This parameter changes the encoding scheme of the TDM bus. The default
type is mu-law. Valid values are a-law and mu-law.
Starting Media Port Starting port for AudioCodes RTP sessions. The default value is 4000. If this
port conflicts with other resources or applications then set this parameter to
change the starting port. This value must be divisible by 10. AudioCodes
port assignments increment in pairs of three from the starting port and
consecutively to the number of IP resources * 10.
For example, if the starting port was 4000, then the first IP resource will
consume 4000 (for RTP), 4001 (for RTCP) and 4002 (for T.38 fax). The next
IP resource will consume 4010, 4011, 4012 and so on. If this was a 120 port
card, the last IP resource will consume 5190 (for RTP), 5191 (for RTCP) and
5192 (for T.38 fax).

Minimum Jitter Buffer In milliseconds. Minimum jitter buffer delay that will be used by the
Delay dynamic jitter buffer algorithm on AudioCodes boards. The algorithm will
never reduce the jitter buffer below this value.
Values: 0..150 (milliseconds), 40 is the default
Considerations: Consider what minimal delay would be safe over a low jitter
network and set the AudioCodes Minimum Jitter Buffer Delay to that minimal
value.
The optimization factor should be governed by the application’s relative
sensitivity to packet errors and delay. Set a high optimization factor if the
application is sensitive to packet loss and a low optimization factor if it is
preferred to pay for low delay with a higher error rate.
Jitter Opt Factor Jitter buffer optimization factor is a unit-less value that determines the
operational response of the dynamic jitter buffer algorithm on AudioCodes
boards. If set to the maximum value, the jitter buffer delay tracks the
network latencies to their maximum and stays there, thus minimizing packet
loss but maximizing delay. When the lowest value is used, the jitter buffer
increases delay only to compensate for clock drifts, and soon decays to it
minimal setting again, thus minimizing delay but maximizing packet loss.
Values: 0..12 (7 is the default)
Selecting Board configuration will bring up this dialog.
MAC Address 12-digit MAC address of the board or 0 (zero). The MAC address should be
entered as shown on the sticker attached to the physical card.
Master Whether the board is the clock master for the bus or clock slave,
respectively. If your system contains any Dialogic or Aculab boards then this
value will always be unchecked (slave).
H.100 Termination Whether the board should terminate the H.100 bus. If the board is situated
on is the first or last board on the bus then hardware termination should be
checked. Otherwise, if the card is between Dialogic or Aculab cards then
H.100 termination should be unchecked.

IP Address IP address/subnet mask assigned to the board. Enter it in dotted decimal

format, for example, 10.12.1.15. WARNING: This IP address must not be
Subnet Mask the same as the IP address of the network card in the Interaction Center
Server. Each AudioCodes board will have its own unique IP address.
Default Gateway IP address of the default gateway machine. Enter it in dotted decimal
format, for example, 10.12.1.1. If you do not have a default gateway, use
the IP address of the host NIC (i.e. the IP address of the Interaction Center).
Server This parameter is for documentation purposes only. It is useful for

switchover (in switchover, you configure all the board in both servers) to
document the board placement.
Port Duplex AudioCodes defaults to 100Mbps half-duplex if the switch port is not set to
auto-negotiate. If the negotiation fails (which happens if the switch port is
not configured for auto-negotiate), the AudioCodes board will drop down to
its default setting of 100Mbps half-duplex. If AudioCodes is running at half-
duplex and the switch port is at full-duplex, packets will be dropped and
audio will be choppy.
18 Installing and Configuring Intel HMP Software Solution

Use this section if using Intel HMP (Host Media Processing). This is a total software solution.
18.1 Important Notes and Restrictions

HMP 1.3 is the supported release for IC 2.4. HMP 1.1, which is used for IC 2.2.x and 2.3.x, is not
supported with IC 2.4.
An overview of Intel HMP product is available on the Intel web site.
18.2 Servers
The Interactive Intelligence we site maintains a list of servers for certified for use with AudioCodes
boards.
18.3 Densities
These limits apply to HMP 1.1. There are no hard limitations for HMP 1.3, but the general rule is that
1.3 numbers are double the number of 1.3.

Letter HMP Resources HMP 1.1 Notes
in Limitations
xml
file Note that HMP 1.3
name approx. doubles
these numbers.
Total number of Total <= 254 The sum of all resources.

resources
AND
R+V+F+C/2 <= 254
AND
R+V+C <= 254
R RTP G.711 Resources R<= 120 The number of RTP resources for any given
(i.e. G.711 IP configuration should be greater than or equal
Resources) AND to the number of voice, conferencing, or fax
R>= max (V,C,F) resources (whichever requires the highest
number of resources).
E Enhanced RTP E<= 64 This is the number of RTP G.711 resources

resources (Adding (above row) that are capable of low bit rate
G.723/G.729 feature to AND coding (G.723/G.729).
the RTP G.711 E<= R
Resources)
V Voice Resources V<= 120 These are used for play, record,…. In 2.3.1,
transaction record is available.
AND
V<= R
S Speech Integration S<= 120 This is the number of voice resources (above
Resources row) that will have CSP (continuous speech
AND processing) capabilities. CSP is used to
S<= V provide an echo cancelled stream of audio
for ASR (i.e. speech rec). You will need a
CSP resource for every session that is doing
ASR (in other words, any call where the
AsrBeginSession tool step has been executed
but has not executed the AsrEndSession tool
step.
C Conferencing C<= 120 Note that the maximum parties in a

Resources conference is equal to the number of
AND conferencing resources you have on your
C<= R system. For example, if you have 100
conference resources, you can have 1
conference with 100 parties.
F T.38 Fax Termination F<= 32 Number of T.38 IP sessions terminating into

Resources the IC server.
AND
F<= R
IP Call Control NA Always 0. This is not needed since

Interactive Intelligence’s SIP stack is used
for the call control
18.4 Vendor Software

The Intel/Dialogic HMP software:
• Install Intel HMP Release 1.3 and the necessary Intel Service Packs and PTRs, which can
downloaded from http://www.inin.com/support/dialogic/software/index.asp?

• Contact Interactive Intelligence for the HMP license file. The HMP license file MUST be
acquired through Interactive Intelligence for support. See
http://www.inin.com/support/dialogic/software/index.asp? for information.
18.5 Configuring your HMP system.

Make sure you activate your HMP license. The license file activates virtual boards, that contain virtual
resources.
18.5.1 Service Setting

Intel HMP should be set to semi-automatic mode (see the Intel Hardware Application Note for details
on each mode). Semi-Automatic is “Settings” | “System/Device autostart” | “Start Server Only”.
Note that “Detect Device” is the new Intel wording for “Manual” and “Start System” means
“Automatic”.
18.5.2 QoS Setting

Also see section 10.1.3 “Notes about QoS and the Interaction Client.”
There is a registry parameter DisableUserTOSSetting that is used by Windows OS and it prohibits the
Windows IP stack from accepting the TOS value from the application. This parameter is ON by default.
To use the QoS setting (in the audio page in the line, station, and global station), a windows 2000
registry setting from http://support.microsoft.com/default.aspx?scid=KB;en-us;248611& must be set.
18.5.3 IP addresses
The IP address is configurable via the DCM. Select HMP Software on the system tree view in the DCM
window, select configure device, and the “Default IP Address” Tab.
The old, retired method used in HMP 1.1 was:
The IP address is configured when you install HMP. If you change your IP address, you must
reinstall HMP or go into the registry and change it. It can be found at
HKEY_LOCAL_MACHINE/Software/SBLabs/dm3ssp.
18.5.4 Timers
HMP requires a high resolution timers for real time processing of 10, 20, and 30 millisecond frames.
There are two timers that can be used:
• Microsoft Windows Multimedia Timer (mmtimer). This is a software timer. This timer is on
higher speed machines (1GHz and beyond).
• Advanced Programmable Interrupt Controller (APIC). This is a hardware timer (via an on-chip
controller on the Pentium family processors). This is more accurate than the mmtimer.
Starting with new Intel releases (HMP 1.1 SU 20 and beyond), the Intel HMP install program will
disable the APIC timer if necessary (and will use the mmtimer) and put up the follow dialog:

For older releases, you will have to configure the timers ‘manually’. See below.
Problem 1 The local APIC’s operation may not be reliable when used in conjunction with some
chipsets if Advanced Power Management (ACPI) is installed.
Solution Run dialogic/bin/readfadt.js to see if there is a conflict with the APIC timer and other tasks, such
as Advanced Power Management. If there is a conflict, then you have 3 options:
Option 1 Stop the dlgcapidrv service, thus causing the mmtimer to be used.
How: Bring up the DCM dialog (keeping the Intel/Dialogic service stopped). With the DCM dialog
up and Intel/Dialogic service stopped, use a command prompt to run the command “net stop
dlgcapicdrv”. Start the Intel/Dialogic service with the DCM
Note: Option 1 needs to be done EVERYTIME you start the Intel/Dialogic service.
Option 2: Disable the dlgcapidrv service, thus causing the mmtimer to be used.
How: Change the value of
HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\DlgcApicDrv\Start from 2 to 4 (this
will cause the dlgcapicdrv service to be disabled). Unfortunately, it doesn’t show up in SCM, so
you have to do it through the registry.
Note: Option 2 only needs to be done once.
Note: You must restart Windows for this registry setting to take affect.
Option 3 Disable the Advanced Configuration and Power Interface (ACPI), thus resolving the conflict so the
Advanced Programmable Interrupt Controller (APIC) can be used.
How: Using Microsoft KB article #Q237556
(http://support.microsoft.com/default.aspx?scid=KB;en-us;q237556), change the computer type
to “Standard PC”.
18.6 Configuring RTP Dynamic Port Range

HMP currently defaults UDP port usage from 49152 to 49XXX for RTP streaming, where 49XXX=49152
+ twice the maximum number of channels purchased under the HMP licensing agreement. For
example, if 120 channels were purchased, 49XXX would be 49392.
If the UDP/RTP port rang used by the HMP system conflicts with other RTP services such as firewall
configuration, the steps below can be followed to set a different range:
1.) Stop the DCM service

2.) Locate the .config file in the c:\Program Files\Dialogic\data directory that matches the
FCD/PCD files associated with you licensed configuration.
3.) Using a text editor open the .config file.
4.) In the [IPVSC] section, go to line “setparm 0x4005, 49512 !set the rtpPortBase on IPVSC”.
The default number is 49152 for this parameter. Change it to what’s desired and save the
file.
5.) Open the Command Prompt window and go to c:\Program Files\Dialogic\data.
6.) Execute fcdgen as follows:
…\bin\fcdgen –f <input filename>.config –o <output filename>.fcd
The resulting FCD file is created in the ..\Dialogic\data subdirectory. If the –o option is
omitted from the command, the default output FCD file will have the same filename as the
user-modified input .config file, but with an .fcd extension.
7.) Restart the DCM service to download the new configuration to HMP.

18.7 Known IC Issues with HMP
The following are the known issues with IC running with HMP.
Issue Enhanced RTP resources are not limited
Description IC assumes that all IP resources are enhanced RTP resources even though some are basic.
Symptom Assume you have 20 G.711 resources and 5 of them are enhanced (can do G.723 and G.729). If
G.729 is selected in IA, the IC server will assume that it can do 20 G.729 sessions. However,
when trying to start the 6th G.729 session, an HMP API will fail and the call will be disconnected.
This is the error message:
Module: IPLinkResource.cpp
Method: CIntelHMPIPResource::QueryDialogicError()
Details: Dialogic IP media library error
Additional Info: ipm_StartMedia() failed for device ipmB1C1 with error Invalid parameter
Workaround Do not use more G.723 or G.729 sessions than what you have on the HMP license.
Or
If using the example of 20 G.711 resources with 5 of them enhanced), create 2 SIP lines:
- 1 line for G.729 only with a maximum of 5 calls
- 1 line for G.711 only with a maximum number of calls of 15
Affected IC 2.3
Fixed IC 2.3.2
Hotfixes
18.8 Known HMP Issues
Issue HMP service does not start automatically when set to automatic with the Services
applet (works when you set to automatically from the DCM).
Description
Symptom None
Workaround This is a problem with all Dialogic releases. Always set the service mode from the DCM OR go to
HKEY_LOCAL_MACHINE\SYSTEM \ CurrentControlSet\
Affected HMP 1.1
Issue Hyper-threading is NOT supported with Windows 2000 on HMP IC Servers.
Description On Windows 2000: HMP 1.1 supports a single processor (without hyper-threading) or dual
processors (without hyper-threading).
On Windows 2003: HMP 1.1 supports a single processor (without hyper-threading), a single
processor (with hyper-threading), dual processors (without hyper-threading). or dual processors
(with hyper-threading).
Symptom None

Workaround Not applicable
Affected HMP 1.1
Issue HMP performance on Windows 2003 is much better than on Windows 2000.
Description Please note that servers running Windows 2003 with HMP far outperformed servers running
Windows 2000. As a result, we are strongly recommending Windows 2003 for servers running
HMP.
Also note that HMP 2.0 will only run on Windows 2003 (Windows 2000 will not be supported with
HMP 2.0)
Symptom None
Workaround Not applicable
Affected HMP 1.1
Issue If a firewall provides an odd port number (via PAT) for RTP, HMP will still transmit on
the even port
Description
Symptom None
Workaround This may well be a Cisco PIX issue – it is unclear from the RTP RFC. Investigating….
Affected HMP 1.1

18.9 HMP Limitations
• HMP does not perform echo cancellation.
• An RTP stream that is no longer being received is not reported. If a phone or gateway where
to reboot and stop transmitting audio then the call would be orphaned and need to be
manually disconnected. SCR #29448.
• G.726 has not been tested by Interactive Intelligence and will not be supported. SCR
#41919.
• The RTP port range could be done via an Intel AIP, rather than going through an Intel .config
file. SCR 41918
19 Creating and Modifying SIP Lines in Interaction Administrator

Interaction Administrator contains configurations in the SIP lines, SIP stations, the Global SIP
Stations. SIP Calls made to and from stations will use the configuration settings in the SIP station,
unless the station specifies “Use Global SIP Station Settings”. SIP Calls made that are NOT from or to
stations will use the configuration settings in the Line.
Notes
• A single SIP line can handle multiple calls.
• When a line is created, the changes take affect immediately.
• When a line is modified, the changes take affect immediately.
19.1 Line Configurations not exposed through Interaction Administrator
Interaction Center Values Description

Parameter (use
DsEditU to set value in
the line configuration)
Modify “RegistrarIPList” “|3600” (default) Registration Interval in seconds.

“|number”
19.2 Creating A SIP Line

In the lines container, right-click in the Lines list and choose Insert New from the menu that appears.
Type the name of the new line, and then select SIP, as shown in the following figure:

Once you click the Next button, the SIP Line Configuration dialog appears, as shown in the following
figure. Complete the fields.
19.3 Line Configuration: Line Page
Line Configuration
Active Same as in today’s Line objects. Note that a SIP line is subject
to being licensed. Only active lines are counted.
Default: On
Phone Number Same as in today’s Line objects. A required field. This number is
used in the “From” header in outbound SIP calls. This value is
not used if a handler changes the origination address in the
Extended Place Call tool.

Line Configuration
Domain Name Domain name used to formulate SIP-URLs for IC users and
phone numbers. This domain name will be automatically
appended to all REGISTER requests sent by the Interaction
Center.
This value is used in the “From” header in outbound SIP calls.

This value is also used in the realm value (in the challenge
request that is sent by IC) in Digest Authentication for station
authentication.
Combined If the Combined radio prompt is selected, the Combined value

prompt is shown.
Inbound/Outbound
If this Inbound/Outbound radio prompt is selected, then both
Inbound and Outbound will be prompted for.
Inbound The maximum number of inbound/outbound calls respectively

that the SIP line will process. When the maximum is reached, no
Outbound more calls will be processed of the exceeded type.
Valid: unlimited, or 0 plus
Default: unlimited
Note: both Inbound and Outbound cannot be 0
Unlimited Button: When this is selected, field will be set to
internally mean the maximum number of calls is unlimited.
Combined When the toggle is checked, the label for the inbound label is set
to “Combined” and the outbound prompts and label are hidden.
Note: this value cannot be set to 0.
When the combined inbound/outbound is reached, no more calls

will be processed.
Disable T.38 Faxing Unchecked (Off) means that T.38 will be used for faxes over
SIP.
Default: Off.
Auto Disconnect when Silence is Same as in today’s Line objects. Linked to Silence Time.
Detected
Default: Off
Silence Time Same as in today’s Line objects

19.4 Line Configuration: Audio Page
Line Configuration
Audio Path Always-In: The audio will flow through the IC server.
Dynamic: The audio will NOT flow through the IC server
whenever possible. The audio will flow though the IC server
only if the IC server determines that it needs access to the
audio. The IC server needs access to the audio for recording,
monitoring, conferencing, and when the two devices don’t have
a common codec.
DTMF Type The type of DTMF signaling. If the connection is to a station, the
DTMF type in the station is used. Possible values:
Inband – DTMF tones are in the actual audio stream.
RFC2833 (default) – DTMF tones are sent and received via tone
information contained in RTP packets. If RFC2833 is selected for
DTMF Type, the Interaction Center server will attempt to
negotiate an audio session with the remote endpoint using
RFC2833 for DTMF, but if the remote side doesn’t support
RFC2833 then it will revert to Inband mode.
RFC2833 Only – will force all sessions to be negotiated using
RFC2833 for DTMF. If the remote side doesn’t support RFC2833
then the session will fail.
Vendor Specific
Intel/Dialogic software (HMP) supports RFC2833.
AudioCodes hardware boards supports RFC2833.

Line Configuration
DTMF Payload The value used for the DTMF RTP payload type. This should be
set to the same value you have configured the other SIP devices
in your network. Many devices do not negotiate the DTMF
payload correctly, so it is very important that each device sets
this parameter to the same value.
Values:
101 (default)
96-127
Vendor Specific
100, 102-105 should not be used for AudioCodes.
This value is also in the station and the global station
configuration.
Network Gain -31 to 31 dB, Default is 0

This value controls the gain applied to the audio signal received
from the IP network. For example, this would be applied to the
signal coming from to an agent’s phone.
Vendor Specific
Intel/Dialogic software (HMP) supports network gain in 2.3.
AudioCodes hardware boards support network gain.
Bus Gain -31 to 31 dB, Default is 0

This value controls the gain applied to the audio signal received
from the TDM bus (and going to the IP network). For example,
this would be applied to the signal going to an agent’s phone.
Vendor Specific
Intel/Dialogic software (HMP) supports bus gain in 2.3.
AudioCodes hardware boards support bus gain.
RTP QOS Byte (hex) QOS byte that will be set in all RTP packets.
Vendor Specific
For HMP platforms, a Windows 2000 registry setting must
also be set.
Voice Activate Detection (VAD) Checked (On) means use VAD on any connection that is NOT to
a station. If the connection is to a station, the VAD configured in
the station is used.
Default: Off
Vendor Specific
Intel/Dialogic software (HMP) does not support VAD.
Echo Cancellation Checked (On) means that echo cancellation will be used.
Default: On.
Vendor Specific
Intel/Dialogic software (HMP) does not support echo
cancellation.
AudioCodes hardware boards support 32ms of echo cancellation
on the voice going from the TDM bus to the IP network.

19.5 Line Configuration: Transport Page
Line Configuration
Transport Protocol Can be UDP or TCP or TLS.

Default: UDP
Note: TLS is not supported in this release.
Address To Use Select the Network Connection (from a drop down list) that you
want to use for all the outbound SIP communication.
Receive Port Port number for which the IC SIP engine will be servicing
requests.
Valid: 1024 to 65535
Default: 5060 for UDP and TCP
Default: 5061 for TLS
Connect Timer (valid for only TCP) TCP: TCP connection time out (in milliseconds). Maximum
amount of time to wait for a TCP connection to be established
before timing out.
Valid: 500 to 20000 (milliseconds)
Default: 3500
T1 Timer (valid for only UDP) UDP: Timer value in milliseconds that represents the initial
incremental delay between packet retransmission.
Valid: 500 to T2 (milliseconds)
Default: 500

Line Configuration
T2 Timer (valid for only UDP) UDP: Timer value in milliseconds that represents the maximum
incremental delay between packet retransmissions.
Valid: 4000 plus (milliseconds)
Default: 4000
Maximum Packet Retry (valid for UDP: Maximum Packet Retry for requests
only UDP)
Valid: 0 to 10
Default: 10
Maximum Invite Retry (valid for UDP: Maximum packet retry for INVITE and ACK requests
only UDP) Valid: 0 to 6
Default: 6
19.6 Line Configuration: Session Page
Line Configuration
Use SIP Session Timer If checked, then, for each active call using the line, IC will send
a SIP OPTIONS message to the remote party’s device to verify
Sip Session Timeout the device and the call on that device is still active. The
OPTIONS message is sent at an interval [in seconds] equal to
the SIP Session Timeout parameter. If the remote party’s
device does not respond to the OPTIONS message then the
Interaction Center will disconnect the call.

Line Configuration
Disconnect on Broken RTP Whether to detect the lack of RTP traffic as a reason to
disconnect the call.
The configuration for the number of seconds to wait before
disconnection is described in section 22 “SIP Telephony
Parameters in Interaction Administrator”.
If no RTP, RTCP, and no comport noise packet (used with VAD)
is received in the configured time, the call will be automatically
disconnected.
Note: If a remote device does, for the time configured, is using
VAD (and not sending RTP) AND does not send comfort noise
packet to initiate the silence, AND does not send RTCP, then this
will be misinterpreted as no RTP and the call will be dropped.
Vendor Specific
Intel/Dialogic software (HMP) does not support this parameter.
AudioCodes hardware boards support this parameter.
Disable Delayed Media When making an outbound call, the media information (using
SDP, Session Description Protocol) can be passed in either the
SIP INVITE message or the SIP ACK message. The remote
party can send its SDP in a SIP 183 response or the SIP OK
message.
Normal Media: The SDP is in the SIP INVITE. An IP resource is
allocated before the INVITE is sent.
Delayed Media: The SDP is in the SIP ACK. An IP resource is
allocated only after the remote party answers the call.
Advantages to Normal Media: Normal Media allows Early Media
(when the remote party sends SDP before the call is answered in
a 183 response) to occur. Early Media is done by gateways to
cut through the audio before the call is completed.
Advantages to Delayed Media: When doing a phone group ring,
IP resources do not have to be allocated to ring a group of
phones. Also, when interfacing with a device that changes RTP
packet codec type on the fly (something that is not supported by
the Interaction Center server), Delayed Media will allow the
Interaction Center to pick only one codec and send that one
codec in the ACK – thus informing the remote party to only use
that codec.

Line Configuration
Terminate Analyses On Connect Checked (On) means, if call analyses is used, to terminate the
call analysis procedure when a SIP connection indication from
the network is received.
Example1: The Interaction Center makes it’s PSTN call via SIP
calls through a SIP/ISDN gateway. This particular SIP/ISDN
gateway only sends a SIP connect message back to the
Interaction Center after the remote party answers the call. If
call analysis is used, you would want to keep checked Terminate
Analyses On Connect, so that call analysis will terminate when
the SIP connect message is received.
Example2: The Interaction Center makes it’s PSTN call via SIP
calls through a SIP/analog gateway. This particular SIP/Analog
gateway always sends a SIP connect message back to the
Interaction Center prematurely, before the remote party
answers the call. If call analysis is used, you would want to
uncheck Terminate Analyses On Connect, so that call analysis
will continue after the SIP connect message is received.
If the connection is to a station, the Terminate Analyses On
Connect configured in the station is used.
Default: On
ASR Enabled Select this checkbox to allocate ASR(Automatic Speech

Recognition) resources on this line.
Default: On
19.7 Line Configuration: Authentication Page

Line Configuration
Line Authentication Use this dialog to enter authentication information for a specific
SIP line. Authentication credentials on the SIP line only apply
to outbound calls from the Interaction Center. SIP line
authentication is only used when a proxy “challenges” an
outbound call.
A SIP line is typically used to send a call to an external party
through a SIP gateway or proxy. If the gateway or proxy
challenges the call with a 401 or 407 response code, then the
User Name and Password on this tab are used to authenticate
the call. The digest access algorithm is used as defined in RFC
2617 HTTP Authentication: Basic and Digest Access
Authentication.
19.8 Line Configuration: Compression Page
See the “IC Regionalization and Dial Plan” tech note for details of compression.

Line Configuration
Codecs See table below.

An ordered list of codecs. The Interactive Center will negotiate
the connection to use the first codec on the supported list. You
can select multiple codecs and then prioritize them by moving
them up or down in the list.
Valid: At least 1 codec must be checked.
Default: G.711 mu-law, G.711 a-law
IA will only store an ordered list of those Codecs that are

checked. The Up/Down buttons are available to order this list.
Also only the G.711 codecs allow the frame size to be modified.
Important: Only configure the codecs the platform supports.
See table below for details.
Name Data Rate Default Possible Default Possible Compressi MOS

Frame Frame Frames Frames on Delay
Size Sizes /Packet /Packet
(Note 3) (Note 4) (Note 5)
Codecs supported by Audiocodes AND HMP.
G.711 mu- 64Kbps 20ms 10,20,30ms 1 1 0.75 ms 4.1

law
G.711 a-law 64Kbps 20ms 10,20,30ms 1 1 0.75 ms 4.1
G.723.1 6.3 6.3Kbps 30ms 30ms 1 1,2,3,4 30 ms 3.9

Note 2
G.729AB 8Kbps 20ms 20ms 2 1,2,3,4 10 ms 3.7

Note 1 Note 2
Codecs supported only by Audiocodes and NOT HMP.
GSM 061.0 13.2Kbps 20ms 20ms 1 1,2,3,4
G.726 32Kbps 20ms 20ms 1 1 1 ms 3.85
Note 1: Receive: When G.729AB is used, G.729, G.729A, G.729B, and G.729AB can be received. Transmit: If
the VAD checkbox is selected (in station and line config in IA), G.729AB will be sent. If the VAD checkbox is not
selected, G.729A will be sent. The noannexB value in the received SDP is not honored.
Note 2: HMP does not support 4 frames/packet with G.723 and does not support 1 frame/packet with G.729.
Note 3: Data Rate: The data rates shown below do not include packet header overhead. For example, G.711
actually uses 80K-100Kbps. The data rates below are all for half duplex (which is what most conversation are).
However, if VAD is not used, silence is transmitted, thus using double the bandwidth indicated.
Note 4: Packet and Frame size: nice summary on the topic of packet size and frequency from the
www.erlang.com website: "The frequency at which the voice packets are transmitted have a significant bearing on
the bandwidth required. The selection of the packet duration (and therefore the packet frequency) is a
compromise between bandwidth and quality. Lower durations require more bandwidth. However, if the duration is
increased, the delay of the system increases, and it becomes more susceptible to packet loss; 20ms is a typical
figure." So, the more of the voice you put in a single packet (say 60ms versus 20ms) then the more of the voice
you lose if that packet is lost.

Note 5: MOS: The quality of transmitted speech is a subjective response of the listener. A common benchmark
used to determine the quality of sound produced by specific codecs is the mean opinion score (MOS). With MOS, a
wide range of listeners judge the quality of a voice sample (corresponding to a particular codec) on a scale of 1
(bad) to 5 (excellent). The scores are averaged to provide the MOS for that sample.
19.9 Line Configuration: Proxy Page

Line Configuration
List of Proxy Addresses Note: A SIP proxy server is not required, but does provide some
features that might be needed in certain network topologies. A
SIP proxy can do network and also do gateway selection.
Priority list of outbound proxies available to IC product. If an
outbound proxy is configured then all SIP messages will be
indiscriminately sent to it for transmission. All messages will be
sent to the first proxy in the list. The remaining proxy entries
will only be used if the first entry is deemed not operational.
Each entry in the list should be an IP address in the IP4 dotted-
notation or a fully qualified domain name. IA treats this as a free
format field and does little validation. (IP6 notation is not
supported at this time.)
For each IP address, there should be a port. The port number
identifies the port at which the proxy will be servicing requests.
Do not put the Interaction Center Server in this field, since it will
cause all SIP calls to be looped back to the Interaction Center.
Valid: 1024 to 65535
Default: 5060
DNS SRV Currently this feature is not available.
Use tel: Scheme Defaults to “no”. See section 24 “Gateway/Proxy

Configuration”.
19.10 Line Configuration: Registrar Page

Line Configuration
External List List of external telephone numbers that are not configured in our
system but need to be directed to our server when encountered.
Therefore, we must register them with the registrar. Typically,
these are numbers that are provisioned on the PSTN interface
but not provisioned in our system, like a 1-800 number.
IP Addresses Priority list of registrars available for contact registration by the

IC. If a registrar is configured then all IC contacts are sent to it
in a SIP REGISTER message. All messages will be sent to the
first registrar in the list. The remaining registrar entries will only
be used if the first entry is deemed not operational.
Each entry in the list is expected to be an IP address in the IP4
dotted-notation or a fully qualified domain name. (IP6 notation
is not supported at this time.)
For each IP address, there should be a port. The port number
identifies the port at which the registrar will be servicing
requests.
Valid: 1024 to 65535
Default: 5060
19.11 Line Configuration: Access Page
Line Configuration
Granted Access | Denied Address Granted Access: By default, all IP addresses will be allowed
Denied Access: By default, all IP addresses will be denied
Access | IP Address Put the IP addresses in the list. It’s possible to enter a single IP
address or a range of IP address.

19.12 Line Configuration: Region Page
19.13 Line Configuration: Call Putback Page
There are 3 techniques that the Interaction Center server does to transfer calls and release control.
They are:

1. 302 Response: If a blind transfer is issued and the call has NOT been answered, the Interaction
Center server will send a 302 SIP response to the remote device’s INVITE with the number to send
this call to. After the 302 SIP response has been issued, the Interaction Center will have no
knowledge of the call.
2. REFER: If a blind transfer is issued and the call has been answered, the Interaction Center server
will send a SIP REFER message to the remote device with the number to send this call to.
3. REFER WITH REPLACES: If a consult transfer is issued, both calls must be answered at the time
the consult transfer. The Interaction Center server will send a SIP REFER message (with a
Replaces header) to the remote device. Note that a Cisco gateway might need extra
configuration to receive a REFER with the Replaces header. There is configuration information in
the SIP 3rd Party Component Application Note.
Sending of the REFER with REPLACES

Before: Interaction Center has two calls, call1 (caller1) and call2 (caller2).
After: Interaction Center has no calls, caller1 is taking to caller2 independent of Interaction Center
Server.
The ConsultCall tool has two input parameters that are calls. Order of the parameters does matter
for SIP REFERs. The REFER will be send to the second call. Device2 will then Invite Device1. This
hides information about the original transfer target (call2) from the transferee (call1).
Example:
Call1 (remote caller or transferee) is from a gateway to IC server A. Call2 (transfer target) is from
IC Server A to IC Server B. The ConsultTransfer should be issued like ConsultTransfer(C2, C1)
because you want the REFER to be sent to the gateway.
Messages issues when ConsultTransfer(C2, C1) is invoked:
• IC Server A to gateway: REFER message with Refer-To header. The Refer-To header
contains IC B’s address and a Replaces Attribute (which is Call2).
• Gateway to IC Server B: INVITE with a replaces header. The Replaces header contains Call2.
Receiving of transfer messages

Receiving of the REFER with REPLACES
Restriction: The IC server will reject the REFER with Replaces on normal external calls. The
IC server will only accept the REFER with Replaces with both calls (the one with the REFER and
the one referenced by the Replaces) are to configured stations. If allowed on normal
external SIP calls, the IC would be placing a possible call to the target.
Receiving of the INVITE with REPLACES

The INVITE is received with the replaces. The replaced call will be updated with the new IP,
port number, codecs, plus other info for the audio.

Line Configuration
Enable Call Putback Check if you want the Interaction Center to For Call Putback (or
RLT or Release Transfer) to occur, the following must be true:
• The Enable Call Putback checkbox must be checked.
• If using a handler, the BlindTransfer or Consult Transfer
Tool Steps must not specify false for “Use Putback”.
• Both calls must be external SIP calls.
Option Leave this configuration blank.
20 Defining Global Configurations for SIP Stations

The Global configuration is under “Station Space”/”Configuration” in Interaction Administrator. Double
click on “Configuration” in the right panel.
20.1 Global SIP Station Configurations not exposed through Interaction

Administrator
There are not configuration not exposed at this time.

20.2 Global Station Configuration: Addresses Page
Global SIP Note that all these values are used by default by each station. If
Station desired, each station has the ability to individually configure these
Configuration options.
Connection SIP This is the SIP address used to call the SIP device. This address is used by the
Address Interaction Center to initiate calls to this SIP station. It’s also used to send
MWI notifications f MWI is enabled.
Values:
Same as Identification Address. Use the station’s identification address (in the
station configuration) as the contact address. Note: This option should only be
used if the Identification address is a fully-qualified domain name or SIP
address.
Dynamically Updated. Allow the station’s contact information to be
automatically set from SIP URI in the SIP Message Contact Header in the
station’s INVITE or REGISTER SIP message. This option is very useful if SIP
stations use DHCP and can change IP addresses frequently. NOTE: This option
also is susceptible to ‘spoofing’ and can allow a rogue user to masquerade as
another SIP station when not using authentication. If deploying on an unsafe
network then it’s recommended that authentication (section 20.6 “Global
Station Configuration: Authentication Page”) be used to secure the station from
these types of attacks.
In the station configuration, you can also specify a unique contact address if
desired.
20.2.1 Notes on “Dynamically Updated Contact Addresses” and the audio-enabled client.
The Interaction Center client (when running in audio-enabled mode) will use the Microsoft RTC APIs to
send a registration request. Here are two cases.
Case 1.

REGISTER message, station ID=”7104” and notifier set to “aquaman”. Client created
“From” header from the station ID field and notifier setting.
REGISTER sip:aquaman SIP/2.0
Via: SIP/2.0/UDP 172.16.129.124:11320
From: <sip:7104@aquaman>;tag=b8b951f7-cea4-458d-a72d-da850667f0fd
To: <sip:7104@aquaman>
Call-ID: 9e8a6644-136b-4fcc-8813-3ec1d61f6dec@172.16.129.124
CSeq: 2 REGISTER
Contact: <sip:172.16.129.124:11320>;methods="INVITE, OPTIONS, BYE, CANCEL, ACK"
User-Agent: Windows RTC/1.0
Expires: 0
Content-Length: 0
Case 2.
REGISTER message, station ID=”sip:7104@1.1.1.1:5060”. Client created “From” header
from the station ID field.
REGISTER sip:1.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 172.16.129.124:7483
From: <sip:7104@1.1.1.1:5060>;tag=224c331a-cf97-4ecb-9d3e-8207ef618896
To: <sip:7104@1.1.1.1:5060>
Call-ID: e4b2b988-4821-49de-a59c-59500cf61391@172.16.129.124
CSeq: 1 REGISTER
Contact: <sip:172.16.129.124:7483>;methods="INVITE, OPTIONS, BYE, CANCEL, ACK"
User-Agent: Windows RTC/1.0
Expires: 1200
Event: registration
Content-Length: 0
20.3 Global Station Configuration: Audio Page

Global SIP Note that all these values are used by default by each
Station Configuration station. If desired, each station has the ability to
individually configure these options.
All Parameters All these parameters are documented in the Line Configuration
section for the Audio Page.
20.4 Global Station Configuration: Transport Page
All Parameters, except the ones All these parameters are documented in the Line Configuration
below. section for the Transport Page.
Use Proxy for Station Connections Checked indicates that the proxy list configured in the line
configuration in Interaction Administrator should be used to
connect stations. Unchecked the Interaction Center will contact
the stations directly.
Line Group The line (or lines) in this line group will be used to connect to
SIP stations. The line group is configured in the Line Groups
container in Interaction Administrator. This is needed if you
have a large number of lines (for efficiency) or if you want a
specific SIP line to be used for contacting the configured SIP
stations.

20.5 Global Station Configuration: Session Page
All Parameters, except the ones All these parameters are documented in the Line Configuration
below. section for the Session Page.

Station Connections are Persistent Checked indicates that connections to the station are persistent,
and will not be disconnected until the station initiates the
disconnection. Unchecked indicates that when the Interaction
Center determines that the audio path to the station is no longer
needed, the Interaction Center will initiate the disconnection.
Note that if Persistent is used, the number of call appearances
will be 1. The connection will be established by the SIP phone
(when it makes a call) or by the Interaction Center server (when
it calls the SIP phone because a connection is requested via the
Interaction Client (pickup, makecall, listen,…).
Persistent is typically used when the user uses the Interaction
Client exclusively, and does not use the phone to transfer or
consult.
Recommended setting:
Operators: If you want to handle more calls than the phone is
capable (for instance an operator want to handle up to 20
simultaneous calls), check the Persistent checkbox. The
Interaction Client can be used to manipulate a large number of
calls while the phone will be the audio device for the calls. The
phone will show one call while the Interaction Client will be used
to manipulate the calls.
Call Center Agents: If call center agents are using an IP phone
with a headset and using the Interaction Client, Persistent
should be used.
Number of Call Appearances per Enter the number of call appearances the phone can handle. The
Station Interaction Center will send up to the configured number of calls
to the phone.
Note that if Persistent is used, the number of call appearances
will be 1.
Note that if using the Interaction Client, the number of call
appearances should be 1. Why? Because if the phone would
put a call on hold, you can not take it off hold with the
Interaction Client (the phone itself must take the call off hold).
When using 2 call appearances, the phone will put one call on
hold when answering a second call (if over 1 call appearance is
configured).
Recommended Setting:
General: This value should be over 1 for only experienced phone
users
Vendor Specific
Cisco: The Cisco IP phone 7960 can have up to 6 line
appearances (each line appearance is equivalent to a station).
Each line appearance has a unique SIP address. Don’t confuse
line appearances with call appearances. Each line appearance
handles 2 call appearances. Configure the phone to one line
appearance and then this station configuration to 1 or 2 call
appearances.
Pingtel: Pingtel Expressa IP phone has one line appearance that
handles 4 call appearances. Configure station configuration to 1,
2, 3, or 4 call appearances.

Connection Call Warmdown Time For non persistent connection calls, this is the number of
seconds the SIP call will remain active to a station before the
connection call is automatically disconnected. The IC server
could typically send multiple regular calls (for whisper, IVR) to
the station, and having the Connection Call Warmdown Time
greater than 0 will cause the same connection call to be reused
(which is good).
20.6 Global Station Configuration: Authentication Page

Station Configuration station.
If desired, each station has the ability to individually
configure these options.
Station Authentication Use this dialog to enter authentication information for all SIP
stations. If desired, each station has the ability to individually
configure these options. Enabling authentication forces the
phone to authenticate itself with the Interaction Center Server
before the Interaction Center Server processes any request from
the station. SIP station authentication prevents access to
Interaction Center resources from unauthorized SIP devices. If
authentication fails, then the station will not be able to make
outbound calls.
If enabled, the Interaction Center Server will challenge all calls
from phones that match the stations ID address by sending a
401 response. The User Name and Password on this tab are
used to validate the phones response. The digest access
algorithm is used as defined in RFC 2617 HTTP Authentication:
Basic and Digest Access Authentication.
Note: Authentication is only used to challenge inbound calls.

Calls made to the audio enabled client are always from the
Interaction Center server to the audio enabled client.
20.7 Global Station Configuration: Compression Page

All Parameters All these parameters are documented in the Line Configuration
section for the Compression Page.
20.8 Global Station Configuration: Phone Page

SIP
Station Configuration
Phone Manufacturer • Blank (default). This means use the value in the Global
Station, which defaults to “Interaction Client”
• “Interaction Client”
• “Polycom”
• “Generic”
Special Logic:
“Interaction Client” will cause “ININCC=x” tag to be put in the
From Header on an outbound INVITE. x=0 for normal ringing
calls and x=1 for calls that should be “auto off hook”.
“Polycom” will cause the Alert-Info header to be added for
calls that should be “auto off hook”. This feature is available
starting with the 1.1.1 Polycom firmware. The SIP message
Alert-Info headers will look like this:
Alert-Info: <http://localhost/AutoAnswer>
In the Polycom sip.cfg file , set the following alertInfo
attributes: <alertInfo
voIpProt.SIP.alertInfo.1.value="<http://localhost
/AutoAnswer>"
voIpProt.SIP.alertInfo.1.class="3"/>
Phone Model Use only for reference.
21 Creating and Configuring SIP stations in Interaction Administrator

21.1 Creating A SIP Station

Notes
• When a station is created, the changes take affect immediately.
• When a station is modified, the changes take affect when the station is idle (there are no more
calls on this station’s queue). Use the Line Details page on the Interaction Client to verify that
all calls to the station are finished.
Note that each line appearance on a Cisco IP phone has a unique SIP address. Each line appearance
would be configured as a separate station (don’t confuse line appearances with call appearances). It is
recommended to only use one line appearance on a Cisco IP phone. Note a single line appearance on
a Cisco phone can handle 2 call appearances, a single line appearance on a Pingtel phone can handle 4
call appearances, and the Interaction Client can handle an infinite (ok, large) amount of call
appearances (see section 20 “Defining Global Configurations for SIP Stations”).
You can define a SIP station for Interaction Client workstations, stand-alone telephones, and stand-
alone fax machines. To create a SIP station, select the stations container in Interaction Administrator
and press the Insert Key. In the Create New Station dialog, type the name of the new station and click
the OK button. In the station type dialog, select the station type (workstation, standalone phone,
etc.). Click OK.

21.2 SIP Station Configurations not exposed through Interaction Administrator

Parameter (use
DsEditU to set value in
the SIP Station)
21.3 Station Configuration: Addresses Page
SIP
Use Global SIP Station Connection If checked, the values configured in the Global SIP Station
Settings Configuration (see section 20 “Defining Global Configurations
for SIP Stations”) will be used.
Identification SIP Address This is the SIP address that identifies the SIP device. This
address is used by the Interaction Center to identify this SIP
station. See section 21.3.1 “Identification SIP Address Page”
for details.
Connection SIP Address This is the SIP address of the SIP device. This address is used
by the Interaction Center to connect to this SIP station.
Same as Identification Address should rarely be used.
Dynamically Updated will update the connection address when
the phone sends a SIP REGISTER or INVITE.
Other - see section 21.3.2 “Connection SIP Address Page” for
details.
21.3.1 Identification SIP Address Page

When you press the “Identification” button, the following dialog appears.

SIP
Station
Configuration
Identification User This is the SIP address that is used to identify calls made from the SIP station.
and Host When a managed SIP station makes a call, it is routed through the Interaction
Center. The Interaction Center will use this address to identify that the call was
made from a SIP station and then the Interaction Center will complete the call.
Note: The user field is matched against the user portion of the SIP message. IP
phones typically put their configured number in the user portion of the SIP
message. Do NOT confuse this with a user’s extension.
Note: The address of the station when a call comes into the IC server
(identification address) can be different than the address the IC server needs to
use when calling the station (contact address). This is because an inbound call
from a station could be coming through a proxy server. The header of the SIP
message could contain the address of the Interaction Center and not the Station
address itself.
Option 1 “Use a predefined format”, “Use User Portion Only” selected. If running
Switchover, the “Use User Portion Only” must be selected so that each switchover
server can identify the station. NOTE: Using this option without Authentication
can result in “spoofing” and “masquerading” attacks if Authentication is not used.
If the phones are being deployed in environments where this may be a concern,
then it’s recommended that Authentication be used.
Option 2 “Use a predefined format” with the “Use User Portion Only” not selected.
This option provides more security (than option 2) to prevent “spoofing” and
“masquerading” because the IP or host portion of the From address is used to
identify the station as well as the user portion. It’s typically harder to spoof an IP
address as well as a user portion. NOTE: Using this option can still result in
“spoofing” and “masquerading” attacks if Authentication is not used. If the
phones are being deployed in environments where this may be a concern, then it’s
recommended that Authentication be used.
Option 3 “Use an alternate format” should be used rarely.
Important: Setting up a station on the WAN connected over VPN is identical to
configuring a station on the LAN. Note that when a remote device establishes a
VPN tunnel into the network, it is given a local IP address. Since this IP address
can change on every instance of connecting over VPN, the contact address in the
station configuration should be the “name” of the station, rather than the station’s
IP address. For more info on VPNs, see 12.2 “VPN”.
Continued on next page….

SIP
Station
Configuration
Vendor Specific:
Cisco: For Cisco IP phones, the user field is the same value configured for
“line1_name” and the host field is the value configured for “proxy1_address”
(unless using the “proxy_backup” feature on the Cisco phones).
The Identification User and Host value should be (using Option 2)
sip:[value configured for line1_name]@[proxy1_address]:5060
if you are not using “proxy_backup” configuration and not using IC switchover.
“proxy_backup” is used if:
Using 2 or more proxies
Using no proxies and using Interaction Center switchover
Using no proxies and using a local gateway or emergency (911) gateway
The Identification User and Host value when using “proxy_backup” configuration
ro IC switchover should be (using Option 1)
line1_name (no “sip:”, no “@”, no host name, no port number).
This value can only be set using the alternate format of the Identification User and
Host value.
Pingtel: For Pingtel IP phones, the user field is the same value configured for
“PHONESET_EXTENSION” and the host field is the phone’s IP address. The
Identification User and Host value when using Pingtel phones should be sip:[value
configured for PHONESET_EXTENSION]@[phone’s IP address]:5060. See the “SIP
3rd Party Component Application Note” for details (Option 2).
In the above example, the “PHONESET_EXTENSION” would be “7111”, the
phone’s IP address would be “1.1.1.1”, and the identification would NOT be
“2.2.2.2” as the above example shows but would be “1.1.1.1”. Pingtel phones do
put the proxy’s address in the FROM headers of the SIP message (like Cisco
does).
Interactive Intelligence: If using the audio-enable Interactive Client, the
indentification address can be almost anything. The audio enable client will read
this field and pass this to the Interaction Center on REGISTERs and INVITES
Microsoft: For Microsoft Messenger, the user@host portion is the value
configured in Tools | Options… | Accounts tab | Sign-in name. The Identification
User and Host value when using Messenger should be sip:[value configured in
Messenger Sign-in]:5060. See the “SIP 3rd Party Component Application Note” for
details (Option 2 or 3)
In the above example, the PC’s IP address “1.1.1.1”, and the user@host
Messenger “Sign-in name” would be “7111@2.2.2.2.
port The port value is, by default 5060. It is the same value configured in the line
configuration.
21.3.2 Connection SIP Address Page

When you press the “Connection” button, the following dialog appears:

SIP
Station
Configuration
Connection User This is the SIP address of the SIP device. This address is used by the Interaction
and Host Center to connect to this SIP station. This host portion of the Connection SIP
address is the IP address or host name of the SIP device. You should be able to
ping the host address from a DOS prompt on the Interaction Center Server, such
as "ping 1.1.1.1". If you are unable to do so, check connectivity and the host
name or IP address.
In the above dialog, the SIP device has an SIP address of
7105@172.16.128.142. This could have been 7111@SIP001122334455 if
the SIP phone had a host name of SIP001122334455.
port The port value is, by default 5060. It is the same value configured in the line
configuration.

21.4 Station Configuration: Audio Page
SIP
Use Global SIP Station Audio Settings If checked, the values configured in the Global SIP Station
Configuration (see section 20 “Defining Global Configurations
All other parameters If the “Use Global SIP Station Settings” check box is checked,
the values configured in the “Global SIP Station Configuration”
will be used. If the check box is not checked, then these
configurations will be able to be set independently. See the
Global SIP Station Configuration (see section 20 “Defining
Global Configurations for SIP Stations”) for complete
description of these parameters.

21.5 Station Configuration: Transport Page
SIP
Use Global SIP Station Transport If checked, the values configured in the Global SIP Station

21.6 Station Configuration: Session Page
SIP
Use Global SIP Station Session If checked, the values configured in the Global SIP Station

21.7 Station Configuration: Authentication Page
SIP
Use Global SIP Station Authentication If checked, the values configured in the Global SIP Station

21.8 Station Configuration: Compression Page
SIP
Use Global SIP Station Compression If checked, the values configured in the Global SIP Station

21.9 Station Configuration: Phone Page
SIP
Use Global SIP Station Compression If checked, the values configured in the Global SIP Station
21.10 Station Configuration: General Page

SIP
Call Waiting Tone Same as other stations.
Ring Always Same as other stations.
MAC Address FUTURE: Not used in 2.3.
Phone Model FUTURE: Will be used when the different models of the
Manufacturer require different configurations.

21.11 Station Configuration: Appearances
SIP
Appearances For List of stations that have a line appearance that will appear on
this station.
Appearances On List of stations on which a line appearance of this station will

appear.
All other parameters See the help documentation.

Note that special line appearance configuration is needed on
the phone itself.

21.12 Station Configuration: Region
21.13 Station Configuration: Station Options Tab

SIP
Require Forced Authorization Code Same as other stations.
Station Has MWI Light Same as other stations.
Outbound ANI Same as other stations.
22 SIP Telephony Parameters in Interaction Administrator
22.1 Server Configuration: SIP Telephony Parameters Page
SIP Telephony Page Values Description

Configuration
RTP Disconnect Time 1..3600 (default Whether to detect the lack of RTP traffic as a reason to
30). disconnect the call is a check box in the SIP Global
Station, the SIP Stations, and the SIP Line.
This configuration is for the number of seconds to wait
before disconnecting.
Default Display String any string, defaults Used as the SIP display string in the FROM header when
to “Interaction calls are made to persistent SIP managed stations and to
Center” any SIP managed station when the client MakeCall
button is pressed. This string will show on the From field
on the phone display.

SIP Telephony Page Values Description
Configuration
Managed Phone any character Used as a convenient way for managed phones to
Shortcut available from connect to the main IVR.
telephone keypad.
This value should be either the whole SIP address with
type and port number (sip:user@host:port) or just the
See section 21.4 user portion (user).
“Configuring the
Managed Phone Note: This number is typically a “*”. If managed phones
Shortcut” for details. are using a SIP proxy (rather than the Interaction
Center) to make routing decisions, then you must
configure the SIP proxy to route the calls to this number
to the Interaction Center.
Message Button any number Used for voice mail retrieval over the IP phone when the
message button on the SIP phone is pressed.
See section 21.2
“Configuring the This value should be either the whole SIP address with
Message Button for type and port number (sip:user@host:port) or just the
Voicemail Retrieval” for user portion (user).
details.
Note: You must configure voicemail button of the phone
to call this number when it is pressed.
Ringback File Name of wav file, Used to play ringback on external calls when the
defaults to gateway doesn’t not use early media.
“Ringback.waw”
Voicemail Direct any number Used to send calls directly to voicemail for unmanaged
phones. Voicemail for managed phones is already
See section 21.3 handled.
“Configuring Voice Mail
for Non-Managed This value should be either the whole SIP address with
Phones (SIP Diversion)” type and port number (sip:user@host:port) or just the
for details. user portion (user) of the SIP Uri.
The diversion header is used to find what user this
voicemail is destined for.
Note: You must configure your network to send calls,
destined for voicemail, to this number.
22.2 Configuring the Message Button for Voicemail Retrieval

If an IP phone has a message button then it’s possible to configure the phone and the IC server so
that pressing the message button takes the user directly to the voicemail box for message retrieval.
22.2.1 Setup
Set the parameter IP Message Button (see section 22.1 “Server Configuration: SIP Telephony
Parameters Page”).
The Message Button value should be either the whole SIP address with type and port number
(sip:user@host:port) or just the user portion (user).

For example, setting Message Button to “1002” will allow IP phones to configure their message
buttons to dial this number as a convenience for users to retrieve their voicemail. Users can also
directly dial this number if a message button is not available on the IP phone.
The Interaction Center will not ask for a user name and password if the user’s client is active and set
to an available status. You can change this behavior with the Server parameter “Force Message Button
Login”. This defaults to “No” and if set to “Yes” will force users to enter their user id and password.
22.2.2 Vendor Specific
22.2.2.1 Cisco
The parameter for configuring Cisco phones’ message button is messages_uri. See the “SIP 3rd Party
Component Application Note” for details. An example would be 1002@172.16.132.16 where
172.16.132.16 is the Interaction Center’s IP address and 1002 is the value set in the server
parameter IP Message Button.
22.2.2.2 Pingtel
The parameter for configuring Pingtel phones’ message button is PHONESET_VOICEMAIL_RETRIEVE.
See the “SIP 3rd Party Component Application Note” for details. An example would be
1002@172.16.132.16 where 172.16.132.16 is the Interaction Center’s IP address and 1002 is the
value set in the server parameter IP Message Button.
22.3 Configuring Voice Mail for Non-Managed Phones (SIP Diversion)

An unmanaged phone is a SIP phone that the Interaction Center knows nothing about. It doesn’t
have a corresponding SIP station defined for it within Interaction Administrator.
Voicemail is handled automatically for managed phones. If you configure all your phones as stations in
Interaction Administrator, voicemail configuration is already complete. Skip this section.
This section is for phones that are unmanaged phones (phones unknown to the Interaction Center).
These phones (or their proxies) will divert calls to the Interaction Center for voicemail gathering.

22.3.1 Logic
The diverted SIP message will have a request URI that identifies the voicemail pilot number
(sip:voicemail@204.180.46.185) and SIP Diversion header(s) identifying the original destination the
reason for the diversion (Diversion: <sip:5858652@siptest.wcom.com>;reason=no-answer)
INVITE sip:voicemail@204.180.46.185 SIP/2.0
To: <sip:5858652@siptest.wcom.com>
From: "Fred Flintstone"<sip:cic@i3worldcom.com>;tag=26680
Via: SIP/2.0/UDP 204.180.46.185;received=204.180.46.185
Call-ID: 5df33e3731a72d7309a756b4571016d2@204.180.46.185
CSeq: 1 INVITE
Diversion: <sip:5858652@siptest.wcom.com>;reason=no-answer
Notes
• If the CC_Diversion header is received, the Interaction Center treats it as equivalent to the
Diversion header
• If multiple diversion headers are received (or multiple entries in a single diversion header), the
top most header (or first entry) is the last diverted user.
The Interaction Center will set the following values:
For sip address scheme (addresses that start with “sip:”),
type and port number are added if not present in the header
In the above example, Eic_RedirectionTn would be
sip:voicemail@204.180.46.185
Eic_RedirectingTn This is the number that is redirecting the call.

In the above example, Eic_RedirectingTn would be
sip:5858652@siptest.wcom.com
Eic_ReasonForCall “U” for Unknown

“B” for Busy
“N” for No answer
“D” for Direct
“A” for Always
In the above example, Eic_ReasonForCall would be “N”
Eic_ReasonForCallString Exact SIP reason in the Diversion header

In the above example, Eic_RedirectingTn would be no-answer
Notes
• Eic_RedirectionTn contains the whole SIP address with type and port (sip:user@host:port) of
the SIP message URI.
• The handlers check if the VoiceMail Direct parameter equals the whole SIP address
(sip:user@host:port) in Eic_RedirectionTn OR just the SIP address user portion (user) in
Eic_RedirectionTn. If there is a match, the Interaction Center will route the call to the user’s
mailbox whoses extension matches the user portion of the Eic_RedirectingTn, OR to the user
whoses Attribute 2 value matches either the whole or user portion of RedirectingTn.

22.3.2 Setup
Set the parameter VoiceMail Direct (see section 22.1 “Server Configuration: SIP Telephony
The VoiceMail Direct value should be either the whole SIP address with type and port number
Your phones or proxies must be configured to send the calls to the number configured as the
VoiceMail Direct number.
Then configure, in Attribute 2 in the user configuration in Interaction Administrator, the address in the
diversion header, if the address in the diversion header does not match a user extension.
22.3.3.1 Cisco
The Cisco phones do not forward calls to voicemail systems. This responsibility is done by proxies.

22.3.3.2 Pingtel
The Pingtel phones can forward calls to voicemail systems. See the “SIP 3rd Party Component
Application Note” for details.
22.4 Configuring the Managed Phone Shortcut
22.4.1 Setup
Set the parameter Managed Phone Shortcut (see section 22.1 “Server Configuration: SIP Telephony
The Managed Phone Shortcut value should be either the whole SIP address with type and port number
For example, setting Managed Phone Shortcut to “*” or “123” will allow IP phones to dial this number
as a convenience to get to the main IVR for managed phones. Note that the phones must be able to
dial this number (some IP phones do not consider a “*” as a dialed number).
22.5 Configuring Message Waiting Indicators (MWI)

TODO: Set the server parameters Message Light and Message Light Persistent according to section 32
“Server Parameters”.
Insure the voice form on each workstation has “View” | “Control Message Waiting Indicator” selected.
This option is selected from the voice form when a voicemail is opened. This is needed to turn the
MWI off.
Previous to 2.3 RC3, this configuration was in Attribute 3 in the ACD section of the user configuration
was used. Now there is a MWI tab in the user configuration. For station-less users (i.e. users using
Unmanaged Stations) that still require MWI, this value in the user configuration of IA can be used.
The phone number (WITHOUT “sip:” and WITHOUT “:5060”) should be set in Interaction Administrator
| User configuration | ACD Tab | Attribute 3 field (i.e. 2222@172.16.131.11). When voice mail is left

for that user, the Interaction Center will set the MWI on the phone at this address (this logic is in
System_MessageLight.ihd).
Currently, the phone that the client is logged on will light. If no Interaction Client is active, then the
default workstation of the user will light. If no default workstation, then this value will be used. This
logic is in System_MessageLight.ihd and can be changed on a site by site basis.
22.5.1.1 Cisco
The Cisco phones do not subscribe for notifications. The Interaction Center will send unsubscribed
notifications to the Cisco phones.
22.5.1.2 Pingtel
The Pingtel phones can subscribe for notifications. The parameter for configuring Pingtel phones’
subscription is PHONESET_MSG_WAITING_SUBSCRIBE. See the “SIP 3rd Party Component Application
Note” for details. If the Pintel phone subscribes for notifications, the Interaction Center will send
subscribed notifications to the Pingtel phone. If the Pintel phone does not subscribe for notifications,
the Interaction Center will send unsubscribed notifications to the Pingtel phone.
23 Dial Plan Basics for SIP
23.1 Dial Plan General Info

The Phone Number Configuration container in Interaction Administrator accepts SIP addresses for both
Input Conversion and Dial Plans.

For Input Conversions, IC recognizes the following two patterns: "?@Z" (any dialed number containing
an “@” sign) and "sip:Z" (any dialed number starting with the string ‘sip:’). Both of these are
converted to standardized SIP format ("sip:Z") for inclusion in Dial Plans.
Screen shot of the “sip:Z” input conversion:

Screen shot of the “?@Z” input conversion:
Both input conversions above convert the number to “sip:something”. On the Dial Plan page, you can
specify a dial group for handling outbound SIP calls (calls in the format of “sip:something”). See the
Phone Numbers in IC whitepaper (located in the \Documentation directory) for more information on
working with phone numbers and dial plans in IC.

For Dial Group, select the line group that contains your SIP line(s).
Dial Group Configuration
Dial Group The line group with the sip lines to be used for the call.
Dial String The number to be dialed for the specified input pattern. In the above
dialog, “sip:something” for the input pattern “sip:something”.
Important: The trailing “Z” is present to allow “/” dialing and account
code dialing (someone might dial 201-555-1111/123).
23.2 Dial Plan Verification and Testing

You can verify your changes to the dial plan by selecting the “Simulate Call” tab, enter the phone
number, and select the “Simulate Call” button.

24 Gateway/Proxy Configuration
There are three ways to get a call to a gateway or a proxy (explicitly, configuring the gateway in the
Proxy section, or using Dial Plan):
1. Explicitly: Dial number@gateway (i.e. 5551212@10.0.0.90). The call will be sent to the
gateway (at 10.0.0.90) via SIP. The gateway will then in turn dial out 5551212 to the PSTN network.
Notes: Dialing explicitly is cumbersome to dial and forces the dialer to pick the gateway.
2. Recommended: Configuring the gateway in the Proxy section of the line: In the proxy
section of the line configuration, put the gateway (or proxy) as the proxy. All calls will be sent to the
proxy. This allows you to NOT have to do add any “@gateway” translations to the dial plan. See
the table below for translations.
If a Proxy is configured in Line Config… If a Proxy is not configured in Line
Config
Number is in a non tel or TsServer will convert the number to a tel TsServer will convert the number to
non sip format, i.e. format. i.e. tel:3178723000. a tel format. i.e. tel:3178723000.
3178723000. SIPEngine will fail the call since
there is no IP address to send the
If “Use tel: Scheme” is checked, SipEngine call to.
will reconvert the number from tel to a sip
format (sip:3178723000@proxy) and send
the call to the configured proxy.
If “Use tel: Scheme” is not checked,
SipEngine will keep the number in the tel
format (tel:3178723000) and send the call to
the configured proxy.
Number is in a sip format, No conversion by TsServer or SIPEngine, the No conversion by TsServer or

i.e call will be sent to the configured proxy. SIPEngine, the call will be sent to
sip:3178723000@10.0.0.90 the IP address in the number, ie.
10.0.0.90.
3. Use Dial Plan: In the dial plan (described in this sections below), configure a translation from
5551212 to sip:5551212@gateway.
Directions of to configure the DialPlan are below.

Important: No configuration of the DialPlan are needed if you configure the gateway (or proxy) with
the proxy above.

24.1 Dial Plan: Configuring Gateway Selection
Two examples follow on the next pages.
In this first example, a gateway (called “gateway1”) is used for every call to the 201 area code.
Ordinal syntax (i.e. “{7}”) is used since wildcards (NXYZ?) can not be mixed with alpha characters (i.e
“gateway”) in the dial string.
Dial The line group with the sip line to be used for the call. This line group (“SIP Lines” in the above
Group dialog) will typically contain one SIP line.
Multiple Gateway Note when using a single SIP Line: Add a second dial group entry when using
multiple gateways. This second entry will be used if the first entry fails. The second dial group
entry will have a dial string equal to the 2nd gateway’s name or IP address. The same SIP line
group can be specified on both entrys. When using a single SIP Line in the two entries, you are
using the SIP message responses from the gateway to inform the Interaction Center that the
gateway is congested.
More complicated and rarely used - Multiple Gateway Note when using multiple SIP Lines: Add a
second dial group entry when using multiple gateways. This second entry will be used if the first
entry fails. The second dial group entry will have a dial string equal to the 2nd gateway’s name
or IP address. When using a multiple SIP Lines in the two entries, you are using both the SIP
message responses from the gateway and the number of calls configured in the line
configuration to restrict the number of calls sent to a particular gateway.
Dial String The number to be dialed for the specified input pattern. In the above dialog, the number to be
dialed is 1201Nxxxxxx@gateway1 for the input pattern +1201NxxxxxxZ.
Important: Ordinals are used (i.e. “{7}”) rather than the wildcard syntax (NXYZ?) since the
wildcard syntax (NXYZ?) can NOT be used with alpha characters, such as “gateway1”. A
wildcard syntax is shown below.
Important: Specify the dial string with the Dial Group rather than in the “Default Dial String”
field. The default dial string field is only used when no dial groups are specified.
Important: The trailing “{13}” is present to allow “/” dialing and account code dialing (someone
might dial 201-555-1111/123).

In this second example, a gateway (at IP address 172.16.128.4) is used for every call to the 202 area
code. Wildcards (NXYZ?) can be used since there aren’t any alpha characters in the dial string.
Dial Group The line group with the sip line to be used for the call. This line group (“SIP Lines” in the above
dialog) will typically contain one SIP line.
Multiple Gateway Note when using a single SIP Line: Add a second dial group entry when using
multiple gateways. This second entry will be used if the first entry fails. The second dial
group entry will have a dial string equal to the 2nd gateway’s name or IP address. The same
SIP line group can be specified on both entries. When using a single SIP Line in the two
entries, you are using the SIP message responses from the gateway to inform the Interaction
Center that the gateway is congested.
More complicated and rarely used - Multiple Gateway Note when using multiple SIP Lines: Add
a second dial group entry when using multiple gateways. This second entry will be used if the
first entry fails. The second dial group entry will have a dial string equal to the 2nd gateway’s
name or IP address. When using a multiple SIP Lines in the two entries, you are using both
the SIP message responses from the gateway and the number of calls configured in the line
configuration to restrict the number of calls sent to a particular gateway.
Dial String The number to be dialed for the specified input pattern. In the above dialog, the number to be
dialed is 1202Nxxxxxx@172.16.128.4 for the input pattern +1202NxxxxxxZ.
Important: Ordinals (i.e. “{7}”) are not used in this example. The wildcard syntax (NXYZ?)
could be used since there are no alpha characters, such as “gateway1”.
Important: Specify the dial string with the Dial Group rather than in the “Default Dial String”
field. The default dial string field is only used when no dial groups are specified.
Important: The trailing “Z” is present to allow “/” dialing and account code dialing (someone
might dial 201-555-1111/123).
24.2 Dial Plan: Configuration of Displayed Numbers

You can change what portion of the SIP address (for inbound and outbound calls) you want displayed
on the client.

24.2.1 Example 1
The example below, a new Dial Plan object “sip:NxxNxxxxxx@Z” was created so only the user portion
of the SIP address (({5}{6}{7} {8}{9}{10}-{11}{12}{13}{14}) is the ordinal of the user portion)
is displayed.
A sip inbound call from sip:3178723000@sip.inin.com will be displayed as (317) 872-3000.
24.2.2 Example 2
The example below, a new Dial Plan object “sip:?@Z” was created so only the user portion of the SIP
address ({5} is the ordinal of the user portion) is displayed.
A sip inbound call from sip:marketing@sip.inin.com will be displayed as marketing.
24.3 Multiple Gateway Configuration

When using multiple gateways, there are two concerns:
• Detecting gateway failure and/or congestion.

• Choosing the proper gateway to do the outbound call, and being able to ‘roll’ to another
gateway with failure or congestion is detected.
24.3.1 Detecting Gateway Failure and/or Congestion

Current methods to detect unreachable gateways are:
• ICMP (not used by the Interaction Center)
• SIP Timers (used by the Interaction Center)
Current methods to detect gateway congestion and other gateway level errors are done with SIP
response codes:
SIP Response Codes (used by the Interaction Center). Server level errors (SIP errors in the 5xx
range) are re-tryable.
24.3.2 Configuring Gateway Selection by using an External Proxy

Configure the Interaction Center Line’s configuration to have a proxy. The Interaction Center will then
forward the call to the proxy and the proxy will forward the call to the correct gateway. Detecting
unavailable gateways will be done by the proxy.
24.3.3 Configuring Gateway Selection by DialPlan

Configure the Interaction Center Dial Plan to translate the dialed number to sip:number@gateway.
This is described earlier in this chapter.
25 Call Analysis
25.1 Call Analysis (over traditional connections)

When the lines (T1/E1/ISDN/Analog) terminate into Intel or Aculab boards, call analysis is done with a
mixture of information passed from the PSTN. This information is ISDN cause codes, tones, T1/E1
bits, and analog loop current. Once a remote connection is made, if more analysis is required (to
determine human or answering machine), DSPs on the Intel or Aculab boards are used.
25.2 Call Analysis (over IP)

When a SIP gateway is used with AudioCodes IP boards, call analysis is done with information passed
from the gateway. For example, the gateway can convert ISDN cause codes to SIP reason codes
and pass them to the Interaction Center server in SIP messages. Once a remote connection is made,
if more analysis is required (to determine human or answering machine), DSPs on the Intel or Aculab
boards are used.
When a SIP gateway is used with HMP, call analysis is done with information passed from the
gateway. For example, the gateway can convert ISDN cause codes to SIP reason codes and pass
them to the Interaction Center server in SIP messages. Once a remote connection is made, if more
analysis is required (to determine human or answering machine), HMP DSP-like software running on
the main CPU is used.
Note that tones (ringback, busy, SIT) and voice (human and answering machine) are not received
over a dedicated 64K channel as with traditional digital connections, but received over IP (where some
distortion, packet loss, and delay can occur). Even when using a completely uncompressed audio
codec like G.711 some minor audio anomalies do occur. These are often so subtle they aren't
detectable by the human ear when listening to normal speech. When compression is used, like
G.729, the tones are compressed and decompressed and can not reliably be detected.
The best solution is for the tones to be detected as close to the edge (i.e. the gateway connected to
the PSTN) as possible, and the results passed back to the IC server for processing.

Even when no compression is used (like G.711) and the gateway is passing tones rather than the
results, call analysis done on the tones over an IP connection not be as accurate as with traditional
digital connections.
25.3 Call Analysis (over IP) with Interaction Dialer

Product: Interaction Dialer
Summary: Predictive dialing is only supported via PSTN trunks connected directly to Intel or Aculab
boards in a CIC server. It is NOT supported via PSTN trunks terminated on a VoIP gateway.
Detail: The call analysis and answering machine detection required for predictive dialing are highly
sensitive to noise and delays. Interactive Intelligence can only vouch for results when telephony
trunks (T1/E1 spans) are terminated directly on supported voice processing boards from Intel and
Aculab. Interactive Intelligence has not tested predictive dialing when PSTN trunks are terminated on
VoIP gateways because of the multitude of gateway vendors and configurations. Given the inherent
changes in audio introduced by the conversion of voice to IP traffic, Interactive Intelligence suspects
that call analysis and answering machine detection may be less accurate than with direct terminations
on Intel/Aculab boards and perhaps unacceptably so. For these reasons, Interactive Intelligence
recommends that predictive dialing only be done with direct terminations on Intel/Aculab voice
processing boards.
Outlook: Interactive Intelligence plans to test predictive dialing with certain VoIP gateways and SIP
configurations (especially using Intel HMP software on the CIC server). Also, Interactive Intelligence
is investigating the availability of more intelligent VoIP gateways capable of doing call analysis and
answering machine detection "at the edge" and thus avoiding the pitfalls of doing such work after the
conversion of voice to IP packets.
26 Fax Configuration
Our SIP technologies use the RTP protocol to transport audio across the IP network. Problems occur if
the same technique (RTP) that is used to transport audio is used to transport faxing and modem
communication, especially if compression, packet loss, or network delay occurs.
Even when using a completely uncompress audio codec like G.711 some minor audio anomalies do
occur. These are often so subtle they aren't detectable by the human ear when listening to normal
speech. These are significant to affect audio modulated data carriers (modems and fax).
Even at the slower data rates typically used by fax and modems (9600/14400 baud)
the anomalies in the transport can affect the stability of the carrier.
For faxing, the T.38 protocol solves the IP problem. T.38 encapsulates the T.30 data and handling
the problems that T.30 experiences over IP networks.
Also, HMP does not support T.30, only T.38.
Over IP, T.38 will be the only supported fax protocol.
26.1 Availability
T.38 is available with AudioCodes in CIC 2.2 SR-D and EIC 2.2 SR-A (via HF 1674). It is also
available in all 2.3 releases.
T.38 is available with HMP in CIC 2.2 SR-E. It is NOT available in EIC 2.2 releases. It is also available
in all 2.3 releases. With HMP, faxing is only possible via T.38.

26.2 Fax Detection
Faxes are detected by the Interaction Server via different methods:
• Via the CNG tone. After dialing the called fax machine's telephone number (and before it
answers), the calling Group III fax machine (optionally) begins sending a CalliNG tone (CNG)
consisting of an interrupted tone of 1100 Hz. NOTE: The CNG tone might not be recognizable
as a CNG tone after being compressed and decompressed with low bit rate codecs, such as
G.729 or G.723.
• Via an “to send fax, press start now” option in the IVR. The sender, after dialing but before
sending the fax, can navigate the IVR informing the Interaction Center that a fax is coming.
• Configure a specific number (or group of numbers) as dedicated fax numbers.
• Use the RFC2833 CNG tone. This is currently not supported yet. This method works well with
low bit rate codecs, such as G.729 and g.723.
26.3 Scenarios
26.3.1 Inbound Scenario

• Fax call from PSTN arrives at T.38 capable gateway.
• The gateway will negotiate a standard voice conversation over RTP using whatever codec would
have been chosen for a voice call. Note that if the codec is a low bit rate codec, such as G.729
or G.723, and your are relying on the CNG tone for the Interaction Center to determine that
the call is a fax call (rather than DID or the IVR option), the tone might not be recognizable as
a CNG tone after being compressed and decompressed.
• The IC server will treat the call as a normal voice call until FaxServer requests a pickup. This
normally occurs at the root level of the IVR once the IC server hears a CNG tone, or fax DID
numbers, or the caller chooses the "To send a fax…" option.
• Before the call is handed to the fax server, the IC server will re-invite the gateway to T.38
mode. NOTE: it is important that the gateway, on inbound calls, does not switch to T38 mode.
It should be instructed to do so by the Interaction Center.
• The gateway can reject the re-invite. For fax resources that require T.38 (as is the case with
HMP), the fax will disconnect; for other resources the call will continue as a voice conversation
would (via RTP). Without T.38, faxing could fail since faxes (T.30) does not work well over IP
(due to latency and packet loss).
• If the gateway accepts the re-invite, the IC server will switch the call into T.38 mode and pass
it to the fax server to begin receiving the fax.
26.3.2 Outbound Scenario

• Call originates from Interaction Center server.
• The IC server will negotiate with the receiving end to use RTP as if it were a voice call.
• If the receiving end knows that the endpoint is a fax device, it will re-invite to T.38
immediately. Otherwise, it will OK the INVITE and continue as if it were a voice call.
• Once the receiving fax device sends a CED tone (instructing us to begin handshaking), the
gateway/receiving end will re-invite the IC server to begin communicating via T.38.
• The IC server will OK the INVITE and stop listening to the RTP audio stream and instead begin
processing the T.38 messages, playing them to whichever device initiated the call
• In the event the gateway/receiving end never re-invites to T.38, the IC server will simply
continue the call as if it were a voice conversation (via RTP). If the codec selected involves
lossy compression, it is likely the fax will fail to transmit. For faxing resources that require

T.38, such as HMP, it will eventually time out, disconnect the call, and log an error indicating
the remote end never negotiated T.38 mode.
26.4 IC Server Configuration

By default, the IC server will attempt to use T.38 whenever possible, so there is no configuration
required to utilize T.38. However, should the need arise to disable it, find the appropriate SIP line in
the "Lines" container in Interaction Administrator. Select "Disable T.38" from within the lines
properties dialog. HMP requires T.38; therefore this checkbox should never be selected on HMP
systems.
26.5 Gateway Configurations

Most gateways will not have T.38 enabled by default, or they will have a proprietary version enabled.
Specific configurations are beyond the scope of this document, but common configurations are
included.
26.5.1 Cisco
Not all IOS versions support T.38, so you should consult Cisco's web site to find which version will
work with your platform. Additionally, during the course of I3 testing, the following versions were
found to have problems with T.38:
12.2(11)T
T.38 can be enabled globally or for each specific dial peer. To enable globally,
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0
To enabled it on a specific dial peer,
dial-peer voice tag voip
dtmf-relay h245-signal
fax protocol t38 ls-redundancy 0 hs-redundancy 0
fax rate 14400
fax relay ecm-disable
session protocol sipv2
For more information about configuring T.38, please visit Cisco’s web site at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/faxapp/t38
.htm
26.6 Potential Issues

As indicated above, faxing over audio codecs such as G.711 is inherently unreliable and is strongly
discouraged. It is recommended that all call legs involved in a fax call utilize T.38. If any one leg of a
fax call remains T.30 over an audio media stream, while it is possible the fax will be received without
issue, it should be expected that failure percentages will increase.
For example, assume a sample site uses a third-party SIP provider directly to a CIC system (Session
1) located at the customer’s site. When a SIP Provider to the CIC system call (Session 1) is
determined to be a fax, another call (Session 2) would be placed to a dedicated fax endpoint, also
located at the customer’s site. For this call, not only would Session 2 need to switch into T.38 mode
(and thus both the IVR server and the Fax endpoint must both support T.38), but Session 1 should
also switch into T.38 mode (and so the SIP provider should also support T.38). Simply making one of
the call legs T.38 will not improve your success rates; it may in fact decrease the success rate as
additional delays are incurred as the IVR transcodes the T.30 over G.711 packets into T.38 packets.

27 Modem Configuration
Our SIP technologies use the RTP protocol to transport audio across the IP network. Problems occur
with faxing and modem communication if compression, packet loss, or network delay occurs.
Even when using a completely uncompress audio codec like G.711 some minor audio anomalies do
occur. These are often so subtle they aren't detectable by the human ear when listening to normal
speech. These are significant to affect audio modulated data carriers (modems and fax). Even at the
slower data rates typically used by fax and modems (9600/14400 baud) the anomalies in the
transport can affect the stability of the carrier.
For faxing, the T.38 protocol solves the IP problem. T.38 is a IP protocol that encapsulates the T.30
fax data and handles the problems that T.30 experiences over IP networks.
For modems, no T.38 equivalent standard has been globally adopted by the community. All such
standards require support on the hardware layer and are therefore outside of the control of the
Interaction Center platform itself. We will be aggressively pursuing support for data modems when
these transports become available, but at the current time, we cannot make any assurances that data
modems will work in a total SIP environment.
28 Tie Line and Multi-site Configuration

By simply adding SIP, connectivity to all other SIP devices on your LAN and WAN becomes available.
This is true of connectivity between SIP IC servers. All SIP IC servers can communicate with each
other.
There are three basic techniques:
• Manual dialing between systems can be accomplished with SIP addressing (the user will dial a
user extension, followed by an “@” sign, followed by the IC server name). For example, a user
with extension 101 on IC server A can dialed by users on server B or C by simply dialing
101@A.
• Tie lines can be configured between systems. A SIP line is no different than a T1 or ISDN line
and can be added to a line group in the very same manner. The dial plan can be configured
to use a line group when dialing a specific number or a specific set of numbers. For example,
when dialing 715-xxxx, the dial plan can be configured to modify the dial string to 715xxxx@B
and chose the line group with the SIP line can be used.
• Multi-site can be configured with SIP lines, just like any other T1 or ISDN lines. Again, in
Multi-site, each system is configured with a set of numbers indicating how to reach each other
system in the collective. So, on IC server A, you would configure xxx@B as the number to
reach IC server B. When someone on server A dials a user extension and that user extension
is on B, Multi-site will dial xxx@B to get to that server. xxx@B will be configured in the dial
plan to use the line group containing the SIP line.
29 Switchover Configuration
Read the Switchover white paper for the most up-to-date switchover information.
29.1 Switchover Component

See the Switchover application note on the Interactive Intelligence web site. In an all SIP
environment, no dataprobe equipment is required. In a mixed trunk line and SIP environment,
dataprobe equipment is still required to switch the trunk lines.
To inform the switchover component that no dataprobe is present, during the install select the “None”
option when it asks what kind of dataprobe is present. If switchover is already configured, you can
also change the “Switch Type” server parameter in Interaction Administrator to the value “None”.

29.2 Audiocodes Configurations
Since the Audiocodes configuration is replicated, configure all boards in both servers. Then, the
configurations for all boards will on both servers. At startup time, the Interaction Center will ignore
the configuration for boards that don’t exist.
For example, on serverA, configure boards A1 and A2 and B1 and B2, even though boards B1 and B2
exist only on serverB.
29.3 Station Configurations

Since the same station configuration on the Switchover Primary Interaction Center is duplicated to the
Backup Interaction Center, it is important that an Interaction Center server’s IP address or host name
does not appear in the station configuration. Make sure that you use alternatives (such as putting
only the user portion of the SIP address in the station’s configuration Identification section). This
information can be found in section 21 “Creating and Configuring SIP stations in Interaction
Administrator”.
29.4 Switchover in a WAN Environment

Often, it is desirable to have a duplicate IC server in a geographically distant location to insure
services could quickly be available in the event of a disaster. Often, the intent of having this duplicate
server is not so much wanting a standby that switches in under 30 seconds, but instead redundant
hardware that could be brought up whenever business was ready to resume. The default switchover
configuration depends heavily on a fast, reliable connection between the two servers; to have it
configured this way in a WAN environment could result in “false positives,” or switchover believing a
problem with the network is actually a problem with the IC server itself, and thus disabling the active
server.
To reduce the occurrence of false positives, sites may want to:
• Run switchover in Manual Mode

• Increase the “Switchover TS Timeout” as it allows more time for a response before indicating
the TS ping failed
• Increase the “Switchover TS Failure Retry Delay” as it gives the network more time to recover
between the first failed TS ping and the confirmation ping
• Increase the “Switchover UDP Maximum Ping Delay” or disable it completely by setting
“Switchover UDP Monitor” to “No”
There is a tradeoff between speed of response and false positives. If you set the values too high, it
could take many minutes before a switch occurs. If you set them too low, a switch may occur simply
because the route between the servers was only temporarily delayed.
If sites keep switchover in automatic mode, keep in mind that if the backup server cannot contact the
active server, it cannot disable it. Also, clients may not be able to connect to the backup server if
their connection cannot make it across the WAN to the opposite server. If a WAN link is severed,
clients on one side may log into one server while clients on the other side will log into the opposite
server. In version 2.3, once the link is reestablished, the backup server will notify the active server
that it is no longer active.
30 Interaction Client Configuration
30.1 Associating the Interaction Client with a Station

The Interaction Client needs to be associated with the SIP phone. First, lets assume that a station with
the name “station1” was configured with Interaction Administrator. This name is not the sip address
configured, but the name of the station. If the client is used on a workstation named the same name
as the name given to the station (i.e. station1), no configuration is necessary. But if the client is used

on a workstation that has a different name, the you must add the “/w=<station name>” to the
Interaction Client.
How:
• Right click on “Interaction Client” | select “Properties”.
• Select the “Shortcut” tab.
• At the end of the string in the “Target” box, add “/w=<station name>” where station name is
the name of the station configured in Interaction Administrator.
30.2 Configuring the Interaction Client for Audio

The Interaction Client can now not do just call control, but also control the audio. No SIP softphone or
hardphone is needed. The Interaction Client does this by interfacing with Microsoft’s RTC Client APIs.
These are the same APIs that Microsoft’s Messenger uses.
Requirements
• Microsoft Messenger. See the “SIP 3rd Party Component Application Note” for details.
Configuration
• Right click on “Interaction Client” | select “Properties”.
• Select the “Shortcut” tab.
• At the end of the string in the “Target” box, add:

• “/mssipaudio” if using a generic audio device (such as headphones)
• “/mssipaudio:ipw” if using the Actiontec InternetPhone Wizard (New in CIC 2.2 SR-C and EIC
2.2 SR-A). Complete configuration of this device is in the SIP 3rd Party Component
Application Note.

• “/mssipaudio:claritel” if using the Clarisys Claritel-i750 (New in CIC 2.2 SR-C and EIC 2.2 SR-
A). Complete configuration of this device is in the SIP 3rd Party Component Application Note.
30.2.1 Special Messenger Considerations for SIP Enabled Interaction Client

When running the Interaction Client with the audio option (i.e. the /mssipaudio flag), Messenger must
have been loaded on the system, but does not need to be active. If you desire Messenger and the
client to run on the same system, there may be a conflict since both are using Microsoft’s RTC Client
APIs. There are three solutions to this conflict:
• Don't run Messenger when running the audio enable Interaction Client.
• OR run Messenger configured NOT to use a communication service. See the “SIP 3rd Party
Component Application Note” for details.
• OR start the Interaction Client before Messenger (Messenger could have or not have a
configured communication service). If not, Messenger will process the incoming calls, thus
blocking the Interaction Client from processing the calls.
30.2.2 Special Server Considerations for SIP Enabled Interaction Client

The Interaction Client with the audio option (i.e. the /mssipaudio flag), should not be run on the
Interaction Server, since Microsoft’s Messenger and the Interaction Center SIP stacks will conflict with
each other.
30.3 Monitoring SIP Line Activity with the Interaction Client

Since each SIP line can have multiple active calls, the Line Details page in Interaction Client is the
best place to monitor SIP line activity.

1. To add the Line Details page in Interaction Client:
2. In Interaction Client, right-click the area to the right of the telephone pages (shown in the oval
in the following figure.) Select Insert page… from the menu that appears.
In the Pages dialog, select Line Details form the Available list, as shown in the following figure.
31 Phone Services
Currently, phone services is only supported on the Cisco 7960/7940 SIP phones. However, this same
mechanism can be used by other phones, but is not included at this time. This mechanism is simply
linking a selection on the phone to a handler, which will then perform the requested feature.
Certain phones have displays that can be used to display menus. These menus can have many of the
same features that the Interaction Client has, such as record call. These displays can also have
custom menus, such as room service, checkout, order food, etc.
By default, the following features are available using phone services with a Cisco 7960/7940 SIP
phone:
1. Log in to (uses the IC Client’s user extension & password)
2. Log out
3. Change user status (Available, Out of Office,…)
4. Call control of current call (hold, transfer, voice mail, record, alert supervisor via email)
All these actions (login, logout, status change, call control) are implemented in the CiscoIPXML.ihd
handler and can be changed to add new features.
Instructions
1. Run the 2.2 CIC SR-C/2.2 EIC SR-A Cisco Phone Services install on a Web Server. The Web
Server must be running Internet Information Services (IIS). This Web Server does not need
to be the same computer as IC Server.
2. Using the Internet Information Services application, make sure that install directory (i3webs by
default) is shared as a web application on the Web Server.
32 Server Parameters
You can set the following values in the Server Parameters container in Interaction Administrator.

Server Parameter
“TsDiag” “AcuConfDisAgcDia TODO: This configuration has moved to the General

g” Telephony page. Adding AcuConfDisAgcDiag to the
TsDiag server parameter will turn off AGC in Aculab
conferences.
“Force Message Button “No” (default) Use to indicate whether the user id and password will be
Login” required at all times. By default, the user id and
“Yes” password are only required if the Interaction Client is not
at an available status.
33 Troubleshooting
33.1 Viewing Call Information

From Interaction Supervisor, you can create a new column in the call window to view the call attribute
Eic_IpInfo, which has information about the call. This information includes codecs, IP addresses, and
more.
33.2 Tracing
The flowing trace topics should be set to a trace level of Notes (61) when debugging TsServer issues.
• TsServer | TsServer – Any TS related investigation.

• TsServer | SIPEngine - All SIP protocol related problems. This contains the SIP protocol
messages and SIP engine state information. This one is very important if debugging problems
with SIP end-devices (phones, gateways, proxies, etc)
• TsServer | SIPUrl - SIP URL problems. This shouldn't be turned on unless directed.
• TsServer | SIPMessage - SIP parser problems. This shouldn’t be turned on unless directed.
• TsServer | AudioHub - All audio related problems.
• TsServer | SIPDebugInCallLog – Turn to s trace level of 100 to activate info messages in the
Interaction Log. These messages can be viewed on a connected call by right clicking on the
call, and selecting “Open object window”. On the Interaction Log tab, information about IP
addresses, ports, whether the audio is internal or external, …is displayed. In 2.3.2, turn to
trace level below 60 (status) to turn off.

33.3 No Audio Problems
• Can you ping the IP board or the HMP server?
• Are you using the Interaction Client with the /mssipaudio flag and Windows Messenger is
active?
• Are all the IP cards configured with unique IP addresses? If they have duplicate addresses,
then one-way audio or no audio will occur (depending on the ARP protocol).
• Are the IP cards configured with the correct subnet mask?

• If using Intel HMP, make sure the IP address of your network card is the same at that in the
registry (see section 18.5.3 “IP addresses”).
• Are you using Microsoft messenger and it’s requesting an odd port number for the RTP
packets?
• Is Dynamic audio on and the remote device can not support reINVITES (turn AudioFlow to
Always-In if this is so).
33.4 Echo
• Echo is a challenging topic to troubleshoot. Make sure you understand echo (see section 10.2
“Echo”) before trying to locate and fix sources of echo.
• The direction of the echo is key to locating it and resolving it. Are the IC users hearing their
own voice or are the remote callers hearing their own voice? If the IC users hears their own
voice then the source of the echo is likely in the remote callers leg of the call.
• Use the RTP Audio Monitor and Analysis Guide to record the audio directly from the network
(see section 34.4 “RTP Audio Monitor and Analysis Guide”)
• Adjust the Audiocodes gain parameters. If the phones are transmitting too “hot”, the agent
could hear their own voice echoed back at them. In this case, the Network Gain can be slowly
turned down to help overcome the phone’s transmit levels.
• Some headphones are susceptible to acoustical echo. Be sure to use I3 certified headsets.
• Some phones generate echo if their volume is turned too high. Be sure to use I3 certified
headsets.
33.5 Audio Quality Problems

• Is your system set to the correct media type (mu-law and a-law)?
• Are your Aculab, Audiocodes, and Dialogic cards set correctly to terminate the H.100 bus?
Older Aculab cards are configured via a switch, the newer Aculab and AudioCodes card are
software configurable. For Dialogic, see each board’s Quick Install Card. For Dialogic IPLink
cards, there is a jumper that must be set.
• Is your clocking is set incorrectly? For example, if you have T1s and IP cards in your system,
you most likely want to derive your clocking from one of the T1 boards.
• Eliminate pieces of equipment in the audio path can determine where audio problems exists.
For example, a phone can be configured to call directly to a gateway (and not through the
Interaction Center server). This could eliminate the IP boards in the IC server to see if the
audio improves on the phone.

• Quality of Service. Networks that carry voice must be configured for Quality of Service (see
section 35 “Voice Issues on Networks ”).
• Check the RTP Sender reports in the TsServer log (see section 10.3 ”RTCP Sender Reports”).
• VAD. Turn VAD off in the line configuration and the station configuration in Interaction
Administrator, and off on the SIP gateway and the SIP phone in question. VAD could cause
problems with the different VAD and CNG protocols. draft-ietf-avt-rtp-cn-06 is not supported
by Dialogic and Audiocodes yet. This draft defines VAD and CNG for codecs (such as G.711
and G.726) that do not explicitly define VAD and CNG. This could cause static (AudioCodes) or
dead air (Dialogic) on the call when there should be comfort noise.
33.6 DTMF Problems
33.6.1 IVR DTMF Recognition Problem

• DTMF, when sent inband, might not retain it frequencies through the network. This is
especially true when compression is used. Out-of-band DTMF (RFC2833) is available. Set the
DTMF type to RFC2833 in the line, station and global station.
• When negotiating RFC2833, the DTMF payload must match. See the section on DTMF payload
in the line, station and global station.
• If using Intel HMP, make sure the IP address of your network card is the same that is in the
registry (see section 18.5.3 “IP addresses”).
33.6.2 No IVR, Plays, or records

• Does this message exist in the TsServer log or the event viewer?
• 15:59:44.453[dc]TsServer:Log.cpp(942):::LogErrorNoAvailableResource():
version=[2.2.039.0] 09/09/02, function=CVoiceDeviceMgr::Reserve(), No Available Resource
Error, details=No available roving voice devices, additional data=<none>
• If so, you have run out of voice resources. The cause is:
• For Intel/Dialogic HMP systems, check how many voice resource are allocated in the
Intel/Dialogic license file.
• For Audiocodes systems, voice resources are used from other Aculab or Intel boards. Make
sure you have configured these correctly.
33.6.3 DTMF from Managed Phone not being recognized by remote system
• See IVR DTMF Recognition Problem above. The phones and Interaction Center should be set
to RFC2833 if possible.
33.7 Miscellaneous
33.7.1 Selecting hold on the Interaction client puts the call in Held, put the IP phone still
shows connected.
• A SIP call can be held by either endpoint, and since the phone did not put the call on hold, it
can not take it off hold (since its side was never held). Thus, the hold state will not show on
the phone. The same goes for a call held by the phone and unheld by the client. It must still
be unheld by the phone for the complete audio path to be connected.
33.7.2 All incoming calls going immediately to held state

• Is the IP address of the IP board set to 0.0.0.0?

33.7.3 External Call made from SIP phone hears IVR rather than making the intended call
Make a call from the phone to an external number. If you hear the IVR then the call is being treated
as a normal inbound call and is not being identified as a call from a managed station. Solution: Check
the “Line Details” page, then verify that the call’s Number field exactly matches the value configured
as the SIP Identification Address of the station in Interaction Administrator (see section 21 “Creating
and Configuring SIP stations in Interaction Administrator”).
33.7.4 Internal Call made from SIP phone is placed correctly, but does not show up on
client.
Make a call from the phone to an internal number. If the call completes correctly but you do not see
the call on the Interaction Tab in the client, then this call is not being identified as a call from a
managed station. Solution: Check the “Line Details” page, then verify that the call’s Number field
exactly matches the value configured as the SIP Identification Address of the station in Interaction
Administrator (see section 21 “Creating and Configuring SIP stations in Interaction Administrator”).
33.7.5 Calls made from SIP phones do not show on Line Details Page
If the call does not show on the Line Details page (see section Error! Reference source not found.
“Error! Reference source not found.”), the SIP message in not making it to the Interaction Center
Server. The Interaction Center must be configured as the phone’s proxy or the proxy must be set to
send outbound calls from this managed station to the Interaction Center.
33.7.6 Phone rings when I use the MakeCall button in the Interaction Client
This is normal. The Interaction Center server must establish an audio path to the SIP phone. This is
accomplished by making a call to the phone. When you make a call from a client, and your phone is
a SIP phone (and not a SIP soft phone running with the audio-enabled client), and you do not have a
persistent connection, then the IC server call the phone.
33.7.7 Managed station not ringing

Check the SIP Contact Address of the station in Interaction Administrator (see section 21 “Creating
and Configuring SIP stations in Interaction Administrator”).
Is the Contact address blank in the station config in IA? Is so, you will see messages like “Station
somestationname does not have a Contact Address”. This can be fixed by:
• Set the contact address to dynamic and it will be filled in when the station sends an INVITE
or REGISTER. Make sure the station has the proxy address configured correctly and that
the ID address in IA matches the line appearance configured on the phone
• Or manually set the the SIP station’s contact address.
Make sure you can ping the host portion of the address. For example, if the SIP address is
fred.flintstone@bedrockgravel.com, insure that is bedrockgravel.com can be pinged from the
Interaction Center.
Make sure the phone supports Delayed Media. If not, check the “Disable Delayed Media” in the station
in Interaction Administrator.
Make sure the
33.7.8 Managed station not ringing
Station capella10790 does not have a Contact Address.

33.7.9 Message Button playing the main menu
Check the number that is configured on the SIP device. It should match the number configured in
section 22.2 “Configuring the Message Button for Voicemail Retrieval”.
33.7.10 Microsoft Messenger window pops for every

incoming call with using the SIP enabled Interaction Client
If using the Interaction Client with the audio option (section 30.2 “Configuring the Interaction Client
for Audio”, the Interaction Client must be started BEFORE Microsoft Messenger. Messenger must have
been loaded on the system, but does not need to be active. If you desire Messenger and the client to
run on the same system, the client should be started before Messenger. If not, Messenger will process
the incoming calls to your station.
33.7.11 “Station Not Reached” error when making calls

from the Interaction Client (when using a SIP station)
The Interaction Center must be able to contact the workstation. See section 30.2.1 “Special
Messenger Considerations for SIP Enabled Interaction Client”.
Verify that this station can be reached. Call if via the station’s Connection Address in IA, ping it,…..
Verify that the station is not set to “Do Not Disturb”. Each SIP station has unique setting for this
configuration.
33.7.12 SIP Address has a “^” in it.

If the address appears like sip:7612^sip.inin.com, chances are that you are routing a SIP call over a
non-SIP line. To verify:
• In Interaction Administrator, under Stations, copy the station’s Connection Address (something
like sip:7612@sip.inin.com:5060).
• In Interaction Administrator, under Phone Numbers, Simulate Call, enter the Connection string
and then simulate the call.
• Make sure a line group with SIP line(s) is selected.
33.7.13 After hitting the Pickup or MakeCall buttons on

my Interaction Client, I still must pick up the handset to answer the call.
This is a SIP limitation. If you select the Pickup, Listen, or MakeCall buttons in the Interaction Client,
the Interaction Center will first call your SIP device to establish an audio path. Typically, your phone
will ring and you must answer the phone (by picking up the handset or hit the speaker button).
There is no way, via the SIP protocol, to make the phone “answer the call”. The Pickup, Listen, or
Makecall buttons on the Interaction Client will stay depressed until you pick up the handset, or the call
times out.
With Microsoft Messenger and the audio enabled Interaction Client: We have solved this limitation by
having the Interaction Client answer the call via Microsoft’s APIs.
With persistent connections: We have solved this limitation by having an audio connection to the
phone stay up continually, until the phone is hung up.
34 Tools
34.1 Command Line Tools

Test response time for ICMP (Internet Control Message Protocol) messages. This is not a real test for
RTP data.

ping –t hostname
See the route the messages take:
tracert hostname
34.2 Coder Bandwidth Usage

http://www.voip-calculator.com/calculator/lipb/
NetIQ
There has been success with users using SIP over their home DSL or cable connection. Listed below
are potential problems with a setup over a cable or DSL connection:
• Most ISPs do not provide QOS. Your voice traffic is NOT guaranteed and either is your
bandwidth.
• For cable connections, upload speeds could be very low, even though download speeds are
high.
• DSL and cable connections are shared (at some point in the network). At 3PM when the kids
come home from school, the extra traffic can compromise your bandwidth.
NetIQ’s Qcheck can test response times, throughput, and streaming
• Qcheck download (registration required): http://www.netiq.com/free/default.asp
• Qcheck Info: http://www.netiq.com/qcheck/default.asp
Instructions
1. Select UPD (on the left side)
2. Select Throughput (on the right side)
3. Load Qcheck on two servers (one server should be the Interaction Center server).
4. Put two servers (one as Endpoint 1 and one as Endpoint 2) in the drop down and get the
results (one server should be the Interaction Center server).
5. Swap the servers in the drop down list and get the results for audio in the other direction.
34.3 Speakeasy
Test upload and download speeds: http://chi.speakeasy.net/
34.4 RTP Audio Monitor and Analysis Guide

Creates PCM files from sniffed network traffic. Great for figuring out sources of echo. Go to
http://www.inin.com/support/sip/files.asp?

35 Index
5.1 Layer 3 Type of Service Byte, 35, 36 Outbound Logic, 43

AudioCodes boards, 45 Overview, 15
installing, 48 Diagrams, 21
Boards Proxy configuration, 87, 89
list of supported, 11, 21, 23, 45 Server parameters, 122
Documentation, 15 SIP
Extenstions Documentation, 15
assigning, 40 Lines, 61
Gateways, 20 Overview, 15
HMP (Host Media Processing) Station configuration, 86
Installing and Configuring, 11, 46, 55 SIP Phones
Installing Handling more than two calls, 21
AudioCodes, 48 Stations, 86
Interaction Client associating with Interaction Client, 118, 119
configuration, 119 SUP Phones
IP Managed Phone Shortcut server parameter, Supported phones, 19
105 Terms, 14
IP Resource Management, 122 Troubleshooting, 123
IPLink Boards Type of Service, 57
configuring in Interaction Administrator, 57, Voice Mail
61 configuration for non-managed phones, 102
installing, 55 Voice Mail Button
type of service configuration, 57 configuring, 100, 101
IPLInk Boards, 45 Voice over IP
Lines, 61 Notes, 30, 31, 35
MWI, 105

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Installation and Reference Guide

Customer Interaction Center®

Last updated 12/1/2005

Always check for a newer version of this document!

Interaction Center Platform Statement

How do I know if I have a documented feature?

SIP Application Note 2 of 129 ©2005 Interactive Intelligence, Inc.

SIP Application Note 3 of 129 ©2005 Interactive Intelligence, Inc.

SIP Application Note 4 of 129 ©2005 Interactive Intelligence, Inc.

SIP Application Note 5 of 129 ©2005 Interactive Intelligence, Inc.

SIP Application Note 6 of 129 ©2005 Interactive Intelligence, Inc.

SIP Application Note 7 of 129 ©2005 Interactive Intelligence, Inc.

SIP Application Note 8 of 129 ©2005 Interactive Intelligence, Inc.

SIP Application Note 9 of 129 ©2005 Interactive Intelligence, Inc.

New for 2.4 12/1/05

SIP Application Note 10 of 129 ©2005 Interactive Intelligence, Inc.

2.1 Interactive Intelligence Web Site

2.2 Third Party Component Certification

2.3 Software Versions and Upgrades

SIP Application Note 11 of 129© 2004 Interactive Intelligence, Inc.

4.1 Known Issues with Interaction Center Products

Issue Double digit detected

4.2 Known Issues with 3rd Party Products

4.2.1 Known Issues with 3rd Party Products (General)

Issue Some SIP devices do not support delayed media

SIP Application Note 12 of 129 ©2005 Interactive Intelligence, Inc.

Fixed A fix is not necessary. Most devices support delayed media

Hotfixes Not applicable.

4.2.2 Known Issues with 3rd Party Products (Microsoft)

Affected RTC 1.1

Hotfixes Not applicable

Affected RTC 1.1

Hotfixes Not applicable

SIP Application Note 13 of 129© 2004 Interactive Intelligence, Inc.

Affected RTC 1.1

Fixed RTC 1.2

Hotfixes Not applicable

4.2.3 Known Issues with 3rd Party Products (Cisco)

Product Cisco ATA 18x 3.1 firmware (all firmware versions).

Symptom Calls for users on Cisco SIP phones can be disconnected.

Hotfixes Not applicable

4.2.4 Known Issues with 3rd Party Products (ActionTec)

Product ActionTec analog to USB adapter

Affected Not applicable

Fixed Not applicable

Hotfixes Not applicable

Unmanaged SIP phone that is unknown to the Interaction Center.

SIP Application Note 14 of 129 ©2005 Interactive Intelligence, Inc.

6.1 Available SIP-Related Application Notes

6.2.1 Other Companies

6.2.2 What is an RFC

SIP Application Note 15 of 129© 2004 Interactive Intelligence, Inc.

6.2.3 SIP Standards

RFC Standards Description

RFC 3261 (replaces RFC 2543) Session Initiation Protocol

RFC 2543bis04 Session Initiation Protocol.

SIP Application Note 16 of 129 ©2005 Interactive Intelligence, Inc.

RFC Drafts Description

draft-levy-sip-diversion-03 Voicemail for unmanaged phones (uses Diversion/CC-

draft-ietf-sipping-realtimefax-01.txt This document describes T.38 faxing call flows using

6.2.5 IP Address and Ports

Protocol IP Address Used Port Number Used