SIP Application Note

SIP Application Note
Installation and Reference Guide
Interaction Center 2.2
Last updated 10/20/2004

(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Always check for a newer version of this document!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/22/telephony
Abstract
This document contains instructions for installing and configuring SIP functionality
on your IC Server.
7601 Interactive Way

Indianapolis, IN 46278
Telephone/Fax: (317) 872-3000
www.ININ.com
Copyright and Trademark Information
©1994 – 2004 Interactive Intelligence Inc. All rights reserved. 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®, Virtual Office powered by the Enterprise Interaction
Center™, Vocalité™, Interaction Administrator®, Interaction Attendant®, Interaction Client®,
Interaction Designer®, Interaction Fax Viewer™, Interaction FAQ™, Interaction Melder™, Interaction
Scripter®, Interaction Server™, Wireless Interaction Client™, InteractiveLease™, and the “Spirograph”
logo design® are all trademarks or registered trademarks of Interactive Intelligence 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, IN 46278
Telephone/Fax: (317) 872-3000
www.ININ.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 159 © 2004 Interactive Intelligence, Inc.

Table of Contents
1 Change Log............................................................................................. 9
2 Where can I get information? ............................................................... 12

2.1 Interactive Intelligence Web Site.................................................... 12
2.2 Third Party Component Certification ............................................... 12
2.3 Software Versions and Upgrades .................................................... 12
2.4 What’s New..................................................................................... 12
2.5 Known Issues with Interaction Center Products ............................. 14
2.5.1 Hot Fixes .....................................................................................14
2.5.2 Known Issues...............................................................................14
2.6 Known Issues with Other Products ................................................. 14
3 Glossary of Terms................................................................................. 16
4 Introduction ......................................................................................... 16
4.1 Available SIP-Related Application Notes ......................................... 16
4.2 Standards........................................................................................ 17
4.2.1 Other Companies..........................................................................17
4.2.2 What is an RFC.............................................................................17
4.2.3 SIP Standards ..............................................................................18
4.2.4 Why has RFC 2543 been replaced with RFC 3261?.............................18
4.2.5 IP Address and Ports .....................................................................19
4.2.6 Security Alert ...............................................................................19
4.3 SIP Q&A .......................................................................................... 20
4.4 Implementation Overview Diagrams............................................... 27
4.4.1 Picture: SIP Hardware Approach Overview .......................................27
5 When is a SIP Proxy Needed ................................................................ 27

5.1 SIP Message Routing ...................................................................... 27
5.2 Phone Specific Routing ................................................................... 29
5.3 When is a Proxy Needed (for the Phone) ........................................ 29
5.4 Gateway Specific Routing................................................................ 31
5.5 When is a Proxy Needed (for the Gateway) .................................... 31
6 Connectivity Overview .......................................................................... 32

6.1 Trunk Interfaces with the Interaction Center.................................. 32

6.2 Station Interfaces with the Interaction Center ............................... 33
7 Connectivity In Detail ........................................................................... 37

7.1 PSTN Connectivity Options.............................................................. 37
7.2 Phone Options................................................................................. 40
7.3 Remote Survivability and Emergency Dialing .................................. 41
7.3.1 Cisco’s NON-SIP SRST (Survivable Remote Site Telephony) ................41
7.3.2 Cisco’s SIP SRST (Survivable Remote Site Telephony)........................42
7.3.3 Interactive Intelligence’s Remote Survivability using SIP ....................43
7.3.4 Emergency (911) Dialing using SIP .................................................44
7.4 Understanding the Audio Path ........................................................ 45
7.4.1 Remote Sites Without Remote Gateways..........................................45
7.4.2 Remote Sites with Remote Gateways...............................................45
8 Typical Sizing ....................................................................................... 46

8.1 IP Resources ................................................................................... 46
8.2 Bandwidth Usage ............................................................................ 46
8.3 Sample Systems .............................................................................. 46
8.4 External Audio Path (in 2.3)............................................................ 47
9 Voice Issues on Networks .................................................................... 49

9.1 Quality of Service (QoS).................................................................. 49
9.1.1 Layer 3 IP Header Byte..................................................................50
9.1.2 Layer 2 Byte (802.1p/Q)................................................................51
9.2 Echo ................................................................................................ 51
9.3 RTCP Sender Reports ...................................................................... 51
10 VPN, Firewalls, Security, and Network Address Translation............... 52

10.1 Security........................................................................................ 52
10.2 Firewalls and NAT ........................................................................ 53
10.2.1 Cisco Firewall Information ...........................................................53
10.3 VPN .............................................................................................. 54
11 Notes About User and Station Extensions .......................................... 54
12 Inbound Logic / DID .......................................................................... 54
13 Outbound Logic.................................................................................. 57
14 Platforms ........................................................................................... 59
14.1 Platform Combinations and Supported Status .............................. 59
14.2 Platform Comparison.................................................................... 59

15 Installing and Configuring AudioCodes Boards .................................. 62
15.1 Important Notes and Restrictions ................................................ 62
15.1.1 Servers ....................................................................................62
15.1.2 Known Issues ............................................................................62
15.1.3 AudioCodes with Dialogic ............................................................63
15.1.4 AudioCodes with Aculab ..............................................................64
15.1.5 a-law and mu-law ......................................................................64
15.2 Prerequisites ................................................................................ 64
15.3 AudioCodes Switch Port Configuration ......................................... 65
15.4 AudioCodes Plug and Play Drivers (wdpnp.sys, ipm260.inf) ........ 65
15.5 Installing the AudioCodes PCI Driver (windrvr.sys) ..................... 69
15.6 Configuring the AudioCodes Boards with Interaction Administrator
71
16 Installing and Configuring Intel HMP Software Solution .................... 75

16.1 Important Notes and Restrictions ................................................ 75
16.1.1 Servers ....................................................................................75
16.1.2 Densities ..................................................................................75
16.2 Vendor Software .......................................................................... 76
16.3 Configuring your HMP system. ..................................................... 76
16.3.1 QoS Setting ..............................................................................76
16.3.2 IP addresses .............................................................................76
16.3.3 Timers .....................................................................................76
16.4 Known IC Issues .......................................................................... 77
16.5 Known HMP Issues....................................................................... 77
17 Creating and Modifying SIP Lines in Interaction Administrator ......... 80

17.1 Line Configurations not exposed through Interaction Administrator
81
17.2 Creating A SIP Line ...................................................................... 81
17.2.1 SIP Configuration Page ...............................................................82
17.2.2 SIP Protocol Page.......................................................................86
17.2.3 SIP Authentication Page..............................................................87
17.2.4 SIP Compression Page ................................................................88
17.2.5 SIP Proxy Page ..........................................................................91
17.2.6 Registrar Page...........................................................................92
18 Defining Global Configurations SIP Stations in Interaction

Administrator ............................................................................................. 93

18.1 Global Station Configurations not exposed through Interaction
Administrator .......................................................................................... 93
18.1.1 Notes on “Allow SIP Regitration” and the audio-enabled client. .........93
18.2 Global Station Configuration Dialog.............................................. 94
19 Creating and Configuring SIP stations in Interaction Administrator .. 99

19.1 Station Configurations not exposed through Interaction
Administrator .......................................................................................... 99
19.2 Creating A SIP Station.................................................................. 99
19.2.1 General Page........................................................................... 100
19.2.2 Connection SIP Address Page .................................................... 101
19.2.3 Identification SIP Address Page.................................................. 102
19.2.4 SIP Authentication Page............................................................ 106
20 Dial Plan Basics for SIP.................................................................... 107

20.1 Dial Plan General Info ................................................................ 107
20.2 Dial Plan Verification and Testing............................................... 110
21 Gateway Configuration .................................................................... 111

21.1 Dial Plan: Configuring Gateway Selection................................... 112
21.2 Dial Plan: Configuration of Displayed Numbers ......................... 115
21.2.1 Example 1 .............................................................................. 115
21.2.2 Example 2 .............................................................................. 116
21.3 Multiple Gateway Configuration ................................................. 117
21.3.1 Detecting Gateway Failure and/or Congestion .............................. 117
21.3.2 Choosing the Proper Gateway: Configuring Gateway Selection by
using an External Proxy.......................................................................... 117
DialPlan 117
22 Fax Configuration ............................................................................ 118

22.1 Availability ................................................................................. 118
22.2 Fax Detection ............................................................................. 118
22.3 Scenarios ................................................................................... 118
22.3.1 Inbound Scenario..................................................................... 118
22.3.2 Outbound Scenario .................................................................. 119
22.4 IC Server Configuration.............................................................. 119
22.5 Gateway Configurations ............................................................. 119
22.5.1 Cisco...................................................................................... 120
23 Modem Configuration....................................................................... 121

24 Tie Line and Multi-site Configuration ............................................... 122
25 Switchover Configuration ................................................................ 123

25.1 Switchover Component .............................................................. 123
25.2 Station Configurations................................................................ 123
25.3 Switchover in a WAN Environment ............................................. 123
26 Interaction Client Configuration ...................................................... 124

26.1 Associating the Interaction Client with a Station ....................... 124
26.2 Configuring the Interaction Client for Audio............................... 125
26.2.1 Special Messenger Considerations for SIP Enabled Interaction Client127
26.2.2 Special Server Considerations for SIP Enabled Interaction Client ..... 127
26.3 Monitoring SIP Line Activity with the Interaction Client............. 127
27 Phone Services ................................................................................ 128
28 IP Resource Management ................................................................ 132
29 Configuring the Message Button For Voicemail Retrieval ................. 134
30 Configuring Voice Mail For Non-Managed Phones (Diversion).......... 135

30.1 Logic .......................................................................................... 135
30.2 Setup.......................................................................................... 137
31 Configuring Message Waiting Indicators (MWI) .............................. 138
32 Configuring the Managed Phone Shortcut ........................................ 140
33 Sample Configurations ..................................................................... 141

33.1 Central Site Only, Primary Interaction Center Only, Cisco IP Phones
141
33.2 Central Site Only, Primary and Backup Interaction Centers, Cisco IP
Phones .................................................................................................. 142
33.3 Central and Remote Site (no remote gateways), Primary
Interaction Center Only, Cisco IP Phones .............................................. 143
33.4 Central and Remote Site (with remote gateways), Primary
Interaction Center Only, Cisco IP Phones .............................................. 145
33.5 Central and Remote Site (with remote gateways), Primary and
Backup Interaction Center Only, Cisco IP Phones.................................. 146
33.6 Cisco IP phone, no Interaction Client (stand alone lobby phone)148
33.7 Microsoft Messenger Soft IP Phone, Interaction Client, User, and
Station................................................................................................... 149
33.8 Microsoft Messenger Soft IP Phone, Interaction Client with Audio,
User, and Station................................................................................... 150
34 Server Parameters ........................................................................... 151

35 Troubleshooting............................................................................... 153
35.1 Tracing ....................................................................................... 153
35.2 No Audio Problems ..................................................................... 154
35.3 Echo ........................................................................................... 154
35.4 Audio Quality Problems .............................................................. 154
35.5 DTMF Problems .......................................................................... 155
35.5.1 IVR DTMF Recognition Problem .................................................. 155
35.5.2 No IVR, Plays, or records .......................................................... 155
35.5.3 DTMF from Managed Phone not being recognized by remote system 156
35.6 Miscellaneous ............................................................................. 156
35.6.1 Selecting hold on the Interaction client puts the call in Held, put the IP
phone still shows connected. ................................................................... 156
35.6.2 All incoming calls going immediately to held state......................... 156
35.6.3 External Call made from SIP phone hears IVR rather than making the
intended call ......................................................................................... 156
35.6.4 Internal Call made from SIP phone is placed correctly, but does not
show up on client. ................................................................................. 156
35.6.5 Calls made from SIP phones do not show on Line Details Page ....... 156
35.6.6 Phone rings when I use the MakeCall button in the Interaction Client
157
35.6.7 Managed station not ringing ...................................................... 157
35.6.8 Message Button playing the main menu ...................................... 157
35.6.9 Microsoft Messenger window pops for every incoming call with using
the SIP enabled Interaction Client ............................................................ 157
35.6.10 “Station Not Reached” error when making calls from the Interaction
Client (when using a SIP station) ............................................................. 157
35.6.11 SIP Address has a “^” in it........................................................ 157
35.6.12 After hitting the Pickup or MakeCall buttons on my Interaction Client, I
still must pick up the handset to answer the call. ....................................... 158
36 Tools ................................................................................................ 158

36.1 Command Line Tools .................................................................. 158
36.2 Coder Bandwidth Usage ............................................................. 158
36.3 NetIQ ......................................................................................... 158
36.4 Speakeasy .................................................................................. 159
36.5 RTP Audio Monitor and Analysis Guide ....................................... 159
37 Index ............................................................................................... 160

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
Updated “Specifying your firmware” section with new table. This now appears on 1/14/2002
page 6.
Typo corrections 1/16/2002
Updated firmware specifications table. Added procedure for changing firmware 1/17/2002
values. Updated DCM Network configuration settings with examples and corrected
values.
Updated IPLink firmware names. 1/21/2002
Added Things to watch for… section with a note about not using Terminal Services 1/23/2002
or Citrix Metaframe to run Dialogic Configuration Manager.
Added related documents to introduction section and added Troubleshooting section 1/24/2002
at the end of the document. Fixed typo is hexidecimal Subnet Mask field description.
Added section at end on monitoring SIP line details through Interaction Client.
More cautions, such as leading 0s in IP address 1/25/2002
Added Configuring Your System For Mu-law section, Notes About User and Station 2/11/2002
Extensions section, Notes About Quality of Service Section, describe the new station
parameters (persistent, call appearances, use proxy), AudioCodes Specific Section,
Sample Configuration section, DID section
More info on configuring call appearances 2/28/2002
Add AudioCodes setup information. 3/1/2002
Add info on SIP addresses ¾/2002
Vendor specific portions, removed terminal services section, add MWI and message 3/18/2002
button configurations, add sip Q&A section, VAD, when changes of stations and line
take affect, add pictures of topologies
Hardware restrictions 3/21/2002
More Q&A charts, Outbound logic section 3/26/2002
AudioCodes update, new AudioCodes boards, new Firewall/NAT section, new 4/14/2002
Identification section in station configuration
VPN, Gateway selection 4/24/2002
Added RTP Sender Report section, better incoming logic description, N+1 and 5/14/2002
redundancy, unique station and user extensions, better info on dial plans
Power Usage, better description of id field 5/28/2002
Update on AudioCodes board model numbers (ver P03), SIP channel bank Q&A. 6/3/2002
On CIC 2.2
GA CD
More version numbers for Aculab, audiocodes firmware path is mandatory, Cisco SIP 7/1/2002
products Q&A, remove retired version P02 AudioCodes boards and ScBus IPM-260A-
120-TIP-CI board, more info on routing (section “SIP Message Routing”), details on
configuring voicemail for unmanaged phones, RFC2833 configuration, configuration
examples in “Sample Configuration” section, better remote site pictures.

Change Date
Edited content for typos. No substantive content changes. (SMS) 7/22/2002
Added EIC release directly by CIC release, new table for hardware platforms, 4.2 SIP 7/30/2002
standards section, misc tools in trouble shoot section, 7.2 VALN info, updated
Dialogic model numbers, more info on trouble shooting DTMF
Diversion header info, when is a proxy needed (chapter 5) 8/13/2002
Added H.100 termination to AudioCodes Setup server parameter, better dial plan for 8/19/2002
gateways
More info on setting 601 Dialogic boards to mu-law (15.3), more info on SIP
standards (4.2), removed ipvs_evr_isdn_net5_311.pcd and 8/30/2002
ipvs_evr_isdn_qsige1_311.pcd from 301 (15.2), fix typo in 15.5 (0x0A should be
0xA0).
More info on makecall button in the troubleshooting section, add more info on
security, attribute 3 for MWI, /NoDataprobe flag for switchover, Bus termination and 9/26/2002
VAD for audio quality problems, updated dates that CIC SR-B fixes are in EIC 2.2
GA, decision tree for “when do I need a proxy”, added known issues section On EIC 2.2
GA CD
More known issues, support for Audiocodes 30 and 60 port boards, Dialogic HMP,
more updates on when a proxy server is needed. 10/18/2002
Updated managed short cut info, large packet size info, reworked known issues
section, firewall config, HMP issues, better diversion documentation, HMP link, better 11/18/2002
doc for switchover and station configuration
Multiple NIC explanation, more work on known issues, tel scheme, more on vad,
HMP fixes, identification for stations. 12/17/2002
Dialogic PTR bundle 1, Audiocodes card placement in Dialogic system, phone

services, whats new section, new /mssipaudio:xxx flags 1/24/2003
More on no audio and hold in trouble shooting section, no IVR trouble shooting,
documented audiocodes switch issue in known issues, better known issues section, 2/28/2003
dial plan config for only displaying user portion of SIP address for inbound and
outbound calls, AudioCodes plug and play PCI drivers
Delayed media, HF 1372 (for CIC SR-C) and 1384 (for EIC GA), repair screen shots
in Phone servers 3/18/2003
New server parameters (AudioCodes Network Gain and AudioCodes Bus Gain) for
Audiocodes (CIC 2.2 SR-C HF 1462, EIC 2.2 GA HF 1163), new hot fix doc for 1462 4/3/2003
and 1463.
In section 14.1 “Platform Combinations and Supported Status”, added the following
qualification to the Intel/Dialogic PCI Hardware and AudioCodes IP Boards 4/4/2003
combination: “Please note that Interactive Intelligence assumes no liability with (PL)
respect to performance under load of the Intel/Dialogic and AudioCodes
combination.”
Ethereal tool (section 33.2.5.5), trouble shooting echo with server parameters
AudioCodes Network Gain and AudioCodes Bus Gain, audiocodes 4.0 firmware, new 5/9/2003
audiocode board part numbers, remove IPLink configuration.
Remote Survivability and redo chapters on connectivity, Disable Delayed Media
config, 2.1 and 2.2 information sections 6/3/2003
Tell me about Cisco’s skinny protocol in the Q&A section, Tie Line and Multi-site
Configuration chapter, section in Audiocodes chapter about switch configuration, 6/16/2003
bandwidth usage, Cisco SIP SRST routers, QoS bytes, multiple gateway selection
Gateway selection (section 22 “Gateway Selection”), new hot fixes 1577 and 1578,
UseOffHookEventForSIPDialing server parameter (section 33 “Server Parameters”), 6/30/2003
new hot fix 1599 and 1601.
New server parameters for Aculab gain control and agc, typos, dsedit parameter
sections, 1633 and 1637 hot fixes 7/28/2003
Registry setting for HMP, global station dsedit parameters, more screen shots for
gateway selection. 7/31/2003

Change Date
Broken RTP Disconnect Time, added T.38 chapter, add info about Audiocodes with
Dialogic boards, server parameters AudioCodes Minimum Jitter Buffer Delay and 8/11/2003
AudioCodes Jitter Buffer Opt Factor, HF SR-A 1670, SR-C 1668, SR-D 1638
New illustrations in Chapter 6: Connectivity Overview

More on the fax configuration chapter, new info about the AudioCodes PnP and PCI
drivers in the AudioCodes chapter. 8/28/2003
Echo in trouble shooting section, more info on Network and Bus gain, more fax info,
disabling secondary clock master 10/13/2003
Dialogic/AudioCodes combo is certified, new features for early media and connection
call warmdown time, always run wdreg_gui install, Eic_OutboundSetupParams 12/11/2003
attribute, modem configuration chapter
New 8.4 section for 2.3 external audio path, Broken RTP Disconnect Time warning
1/20/2004
Combined all gateway selection into a single chapter

3/22/2004
Warmdown time of 0 is wrong, addinged “Inband Transfer Enabled” server

parameter 10/20/2004

2 Where can I get information?
2.1 Interactive Intelligence Web Site

Head support link: http://www.inin.com/support/ has links to supported
platforms and supported releases.
http://www.inin.com/support/cic/22/telephony/docs.asp?q=670&t=TEL&
contains the following documents:
• SIP Application Note: Information about the Interaction Center, Interaction
Center SIP configuration information, and information how to configure the
hardware and software platforms used by the Interaction Center.
• SIP 3rd Party Component Feature Matrix: Information about both certified and
uncertified devices, and what features these devices have. Uncertified
devices have been tested by Interactive Intelligence and certain deficiencies or
lack of market demand are keeping them off the 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: 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

See the SIP 3rd Party Component Feature Matrix on the Interactive Intelligence
Web Site. More info about the SIP 3rd Party Component Feature Matrix can be
found in section 2.1 “Interactive Intelligence Web Site”.
2.3 Software Versions and Upgrades

Get the latest versions of software.
• Interactive Intelligence: The latest releases supporting SIP for each
Interactive Intelligence product. Hot fixes for each release are on the web
site and listed below. You must publish the new handlers that are in IC
service releases (the handlers are not automatically published with you install
service releases).
• AudioCodes: If you are using AudioCodes IP boards, you should install as
instructed in section 15 “Installing and Configuring AudioCodes Boards”.
• Intel/Dialogic Software (HMP): If you are using Intel/Dialogic Software
(Host Media Processing), you should install as instructed in section 16
“Installing and Configuring Intel HMP Software Solution”.
2.4 What’s New

Release What’s New
CIC 2.2 SR-D Multiple Play optimizations. Media platforms (Aculab, Intel HMP, Intel/Dialogic
hardware) can be configured so that regardless of the number of calls into an

Communité 2.2.2 GA ACD queue, only a single play will be used.
Aculab media operations will be spread over multiple threads on a
multiprocessor system.
SIP thread has been improved to use multiple threads to process events.
T.38 for AudioCodes Platform.
Individual gain adjustment per RTP session with AudioCodes Platform.
You can change from delayed media (SR-C new feature and SR-C new default)
to normal media (SR-B default) by only selecting one codec in IA (see section
17.2.4 “SIP Compression Page”) or setting “Disable Delayed Media” (see
section 17.1 “Line Configurations not exposed through Interaction
Administrator”).
HF 1638 required. You can now have the contact address of the stations be
dynamic. See setting “Allow SIP Registration” in the global station
configuration (see section 18.1”Global Station Configurations not exposed
through Interaction Administrator”) and station configurations (see section
19.1”Station Configurations not exposed through Interaction Administrator”).
If using the server parameters “AudioCodes Network Gain” and “AudioCodes
Bus Gain”, these should be removed and the gain parameters in the line,
station, and global station should be used.
HF 1843 required. Early Media. See setting setting “Disable Delayed Media”
(see section 17.1 “Line Configurations not exposed through Interaction
Administrator”).
CIC 2.2 SR-C Phone Services (section 27 “Phone Services”)
EIC 2.2 SR-A Additional support for Actiontec and Clarisys phones (see the SIP 3rd Party
Component Application Note) and section 26.2 “Configuring the Interaction
Communité 2.2.1 SR-C Client for Audio”.
Station authentication configuration in the Global Station Configuration, and the
Station Configuration (required an upcoming hot fix)
Line authentication configuration in Interaction Administrator (requires HF 1372
for CIC 2.2 SR-C).
Delayed media negotiation is used for outbound calls if over 1 codec is
configured in the line configuration (requires HF 1372 for CIC 2.2 SR-C).
Terminate analysis on Connect in the Global Station Configuration, the Station
Configuration, and the Line Configuration in Interaction Administrator.
Audiocodes PCI drivers are installed automatically (section 15.5 “Installing the
AudioCodes PCI Driver”)
Support for “tel” scheme.
More support for diversion (the attributes will be set if the diversion header is
present). Previously, the attributes would only be set if the diverted SIP
message URI matched the IP VoiceMail Direct server parameter (section
“Configuring Voice Mail For Non-Managed Phones (Diversion)”
Use of new draft for REFER (section “Standards).
Integrated the CIC 2.2 SR-B hot fixes.
2.2 CIC SR-B General enhancements
Communité 2.2.1 GA
2.2 CIC SR-A General enhancements
2.2 EIC GA
Communité 2.2 GA
2.2 CIC GA General enhancements

2.5 Known Issues with Interaction Center Products
2.5.1 Hot Fixes
Release Hot Fixes

CIC 2.2 SR-D http://www.inin.com/support/cic/22/updates/indexsrd.asp?q=880
Communité 2.2.2 GA http://www.inin.com/support/cic/22/updates/indexsrd.asp?q=880 (2.2.2 is based
on 2.2 SR-D)
EIC 2.2 SR-A http://www.inin.com/support/eic/22/updates/index2.asp?q=830
CIC 2.2 SR-C You should update to the most current service release above.
Communité 2.2.1 SR-C You should update to the most current service release above.
2.2 CIC SR-B You should update to the most current service release above.
Communité 2.2.1 GA You should update to the most current service release above.
2.2 CIC SR-A You should update to the most current service release above.
2.2 EIC GA You should update to the most current service release above.
Communité 2.2 GA You should update to the most current service release above.
2.2 CIC GA You should update to the most current service release above.
2.5.2 Known Issues

All issues in the most recent releases are issues in previous releases, unless noted.
Issue Workaround Affected Release Hot Fixes

Releases Fixed In
If no *, then
affects this
release
plus prior
releases
CIC 2.2 SR-D
Double digit problem on last CIC 2.2 SR-C
digit on an internal call. The
last dialed digit might be EIC 2.2 GA
taken as the first digits as
well when in the IVR.
2.6 Known Issues with Other Products

Issue Workaround Affected Release Fixed Hot Fixes
Releases In
General

Some SIP devices do You can change from CIC 2.2 SR-C No fix necessary.
not understand delayed media (new Most devices
delayed media. feature and new default) EIC 2.2 SR-A handle delayed
Delayed media is used to normal media media.
for outbound calls (previous default) by
when over one codec only selecting one
is configured in the codec in IA (see section
line. 17.2.4 “SIP
Compression Page”) or
setting “Disable Delayed
Media” to “Yes” in the
Line Config in IA (using
DsEditU) – requires CIC
2.2 SR-C HF 1562, EIC
SR-A HF 1564, CIC 2.2
SR-D, or EIC 2.2 SR-B.
Microsoft
Messenger
Some versions
Microsoft Messenger
will try to use an odd
port number for audio.
This is not valid with
HMP or AudioCodes.
Cisco VPN
Software
Microsoft Messenger Use Microsoft’s PPTP Cisco’s 4.0 VPN.
does not work with VPN software rather
Cisco VPN software. than Cisco’s VPN
software.
The Cisco VPN does
not expose its
interface, thus
Messenger passes
internal IP addresses
in its SIP messages
(SDP and 200 OK).
ActionTec
Actiontec phones only: None. CIC 2.2 SR-C*
A buzz is heard by
remote caller when an EIC 2.2 SR-A*
actiontec phone goes * Actiontec support
offhook to answer a is new in these
call. releases.

3 Glossary of Terms
Managed phone
SIP phone that is configured as a SIP station in the Interaction Center. A SIP
station is configured in the Stations page of Interaction Administrator.
Unmanaged phone
SIP phone that is unknown to the Interaction Center.
4 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 a new SIP interface, Interactive Intelligence is excited to leverage it’s proven
Interaction Center Platform to contact centers, enterprises, e-businesses and
service providers that wish to utilize a SIP-based infrastructure.
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 are not available. Interaction Center
Platform answers this shortcoming by not only adding voice mail, but also a
number of applications.
4.1 Available SIP-Related Application Notes

• Dialogic Application Note. How to install Dialogic 5.1.1.
• 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.

4.2 Standards
4.2.1 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.
Microsoft has not only jumped on the SIP bandwagon, but is now in the first
wagon. Windows Messenger uses SIP for voice, instant messaging and
presence. Windows 2003 Server will include a SIP proxy, registar, 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.
See http://www.cisco.com/warp/public/cc/techno/tyvdve/sip/ for all the
Cisco SIP-enabled products.
SIP is a double edge sword for Cisco. By full support of industry standard
SIP, it will allow their customers a choice of lower cost phones, such as a
free Microsoft Messenger, and a choice of lower cost gateways. Once the
Call Manager supports SIP, the competition will be fierce.
Currently many Interactive Intelligence customers use the Interaction
Center with Windows Messenger and with Cisco phones and Cisco gateways.
The Cisco Call Manager is not needed in a SIP environment.
In addition, most Interactive Intelligence SIP customers use all SIP networks
and do not mix H.323 with SIP. However, Cisco has made public a SIP and
H.323 Integration paper
(http://www.cisco.com/warp/public/cc/techno/tyvdve/sip/prodlit/sh23g_wp.
pdf) that says “While each call control and signaling protocol offers
advantages and disadvantages within different segments of a carrier
network, Cisco solutions make it possible for service providers to use H.323
and SIP in the same network. Cisco has addressed coexistence and
interoperability issues to enable service providers to optimize their networks
and to have the flexibly to meet divergent customer needs.”
This type of direction is very positive, allowing a standard like SIP to
continue to extend customer solutions.
4.2.2 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.
Internet Standards Process: http://www.ietf.org/rfc/rfc2026.txt

4.2.3 SIP Standards
SIP standards are evolving quickly, and the Interaction Center continues to
adhere to the specs for this emerging open standard. Below are the specifications
used. These will continue to changes as the new RFC standards/drafts:
RFC Standards Description

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
Authentication supported. Basic Access has been deprecated
2.2 EIC SR-A, 2.2 CIC SR-C with hot fix by RFC3261 (SIP) and is not supported.
RFC Drafts Description
draft-ietf-sip-refer-02 REFER
2.2 EIC GA, 2.2 CIC SR-B, Communité 2.2.1
draft-ietf-sip-refer-07
2.2 EIC SR-A, 2.2 CIC SR-C
draft-biggs-sip-replaces-01 Replaces
draft-ietf-sip-cc-transfer-05 Consult Transfer (uses REFER/Replaces)
Blind Transfer (uses REFER)
draft-levy-sip-diversion-03 Voicemail for unmanaged phones (uses
Diversion/CC-Diversion)
draft-mahy-sip-message-waiting-02 MWI (uses SUBSCRIBE/NOTIFY)
draft-ietf-sip-service-examples-03 Hold
draft-ietf-sip-events-05 SUBSCRIBE/NOTIFY
Coming soon…. Description
RFC 3261 Session Initiation Protocol, replaces RFC 2543
Features needed in IC 2.2 to be 3261
compliant:
• TCP mandatory
• Via branch id replaces call leg id as the
transaction id
• Url comparison rules were relaxed
• Supported header for extensions
• New route/record-route simplification
draft-ietf-avt-rtp-cn-06 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.
4.2.4 Why has RFC 2543 been replaced with RFC 3261?
The status of RFC 2543 is that it has been obsoleted by RFCs 3261-3266.
These documents mainly clarify and resolve issues and mistakes 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 this document is “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.
4.2.5 IP Address and Ports
Protocol IP Address Used Port Number Used

SIP IP Address of the system’s NIC 5060. This is configurable.
(Network Interface Card)
RTP For the hardware platforms, the IP 4000. This is port number of the first RTP
Address of the IP telephony board.. session. This is configurable. The
For software platforms (HMP), the second RTP session will start at an even
IP Address of the system’s NIC port number higher than 4000.
RTCP For the hardware platforms, the IP This will always be one higher than the
Address of the IP telephony board. port number used for its RTP session.
For software platforms (HMP), the
IP Address of the system’s NIC
Interaction IP Address of the Interaction Center 2633, registered with IANA
Center Notifier system (http://www.iana.org/assignments/port-
numbers ).
Interaction IP Address of the Interaction Center 3508, registered with IANA
Center Web system (http://www.iana.org/assignments/port-
Services numbers ).
4.2.6 Security Alert
There are security alerts for VoIP protocols, H.323 (which is not used by
Interactive Intelligence) and SIP (which is used by Interactive Intelligence).
For H.323: Several critical flaws have been discovered in VoIP products
based on the widely used H.323 protocol:
http://www.cert.org/advisories/CA-2004-01.html
Note that Interactive Intelligence does not use H.323, it has chosen SIP
exclusively as its VoIP protocol.

For SIP, in regards to: http://www.cert.org/advisories/CA-2003-06.html
Interactive Intelligence continues to test its SIP product lines for against
unauthorized privileged access and denial of service attacks.
Details:
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
(http://www.ee.oulu.fi/research/ouspg/protos/testing/c07/sip/ ) 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 reelase 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.
4.3 SIP Q&A

Does EIC, CIC, Communité, Vocalité , Mobilité , Dialer, Recorder, and
all the other Interactive Intelligence products work over SIP?
Yes.
Are advanced features like call monitoring, call recording, call

analysis, music on hold, and all your other features were not possible
with SIP?
Yes they are, if you architect your system with these features in mind. All our
features, like ACD, IVR, Voicemail, are available.
Are these features available today?

Yes. These features are in the CIC 2.2 release, currently available.
Do I have to replace my complete Interactive Intelligence system to

add SIP?
No. In fact, you have two options. One option to add SIP an existing system.
All the existing connections (ISDN, T1, E1, analog phones) can run in the
same server that is running SIP. This allows companies to gradually move into

VoIP. Another option is go 100% SIP. Because we can connect to both
gateways and IP Phones via SIP, you can build a complete, SIP only system.
Does the same Interaction Client work with all these new devices?
Yes. The same Client works with analog phones, SIP hard phones (such as the
Cisco 7960 and the Pingtel Expressa), and SIP soft phones (such as Microsoft
Messenger). In fact, some PBX digital phones are now supported with the Intel
NetStructure PBX-IP Media Gateway.
Does Cisco back SIP? I’ve heard different stories, depending on the
account and salesperson.
From the Cisco web site: “Cisco is enabling the advance of new
communications services with a complete SIP-enabled portfolio including IP
phones and analog telephone adaptors, packet voice gateways, proxy servers,
call control and signaling, and firewalls. These products are available today.”
See http://www.cisco.com/warp/public/cc/techno/tyvdve/sip/ for more info.
Also, Cisco Phone Data Sheets can be found at
http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_shee
ts_list.html. Note that all Cisco phones do not support SIP (yet).
Tell me about Cisco’s skinny protocol (called SCCP), H.323, and SIP.
How do their interrelate?
Here is a little Q&A:
• Why aren't Cisco’s skinny protocol (i.e. SCCP) and other proprietary call
control protocols a standard?
A: In 1997 and 1998, vendors were clamoring for VOIP call control protocols.
Unfortunately, there was not a lightweight call control protocol
available for vendors to standardize upon. As such, vendors such as 3COM,
Cisco, and Avaya each modified the H.323 call control protocols (H.225 and
Q.931 for example) to provide features and functions which would allow them
to compete with existing analog/digital PBX equipment. Because each of
these call control protocols were developed internally at competing
organizations there was never support for standardizing on any one protocol.
• Why would vendors protect their proprietary call control protocols when
the SIP standard is available?
A: Each of the vendors which created a proprietary call control protocol has a
large investment to recoup in order to justify the development efforts of the
protocol. As such, these vendors will be the last to adopt the SIP standard
and open nature that SIP brings to the deployment of VOIP networks. By
continuing to market their non-standard protocols, customers are put into a
very bad position where they must purchase the entire solution from the
vendor.
With SIP's ability to seamlessly integrate network and application components
from multiple parties, the cost justification for deploying a non-SIP based
network is rapidly eroding. There is ample evidence of this in the rapidly
falling cost of phones and network components for a VOIP deployment. Due
to this influence we are seeing the quiet adoption of SIP by the vast majority
of vendors who originally created a proprietary protocol.

• What standards group backs skinny and other proprietary call control
protocols?
A: None, by definition proprietary protocols are rarely ratified by standards
bodies. The standards bodies have backed SIP.
• Are SIP and skinny similar?
A: They are both light weight protocols used for setting up and tearing down
calls. The major difference is that SIP is an open standard that is being
updated to provide communication setup and tear down capabilities not only
for voice but also messaging, video etc.
• When will proprietary VOIP vendors support SIP?
A: As more and more customers demand device interoperability, vendor
choice, and lower prices, proprietary vendors will be forced to support open
standards. Ample evidence of this evolution can be seen through the
sponsorship of SIP Center (http://www.sipcenter.com).
The fact is that many of the vendors who created a proprietary VOIP protocol
support SIP today. Typically the only component that does not support SIP is
the IP-PBX itself since it relies on the proprietary protocol for call control.
• Why would a company looking to deploy a VOIP network purchase a
non-SIP enabled product set?
A: There are multiple arguments in the market place for deploying a single
vendor solution. However, with SIP's ability to utilize products from multiple
vendors and lower the cost of ownership many of these arguments seem to
lose favor.
Show me a stack with Cisco and non Cisco devices and what works
with and without SIP.
The chart below shows that Cisco apps can only use Cisco equipment, require
a Cisco CallManager, and can not take advantage of Cisco SIP phones and SIP
gateways. The Interaction Center with SIP can use Cisco SIP phones and SIP
gateways, PLUS other vendors SIP equipment, and does not need a Cisco
CallManager.
Cisco Apps (Unity, IPCC,…) Interaction Interaction Center with SIP

Center
with the
Cisco TAPI
Platform
Cisco Call Manager
Cisco H.323 gateways and proprietary Cisco SIP gateways and Other vendors SIP
skinny (SCCP) phones SIP phones gateways and phones,
such as Microsoft
Messenger

How does this compare to running the TAPI version of the Interaction
Center?
TAPI vs. SIP comparison TAPI Interaction Center Interaction Center with
SIP capabilities
Can the Interaction Center No. Yes. We simply added SIP

system be mixed with traditional to our telephony board
connections via telephony boards, version.
such as analog phones and ISDN
trunks?
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 – does all the Yes. We have a single No. Multi vendor solutions
gateways, routers, and phones vendor dependency on are used. The first
have to be from Cisco? Cisco. This typically leads gateways and phones we
to higher cost equipment. certified are from Cisco.
We have now certified
phones from Microsoft
and Pingtel. Our system
can work with any
certified SIP compliant
gateway or phone.
Software – are proprietary or Proprietary. Mixture of All SIP.

standard protocols used to standard and proprietary
communicate with phones and protocols.
gateways.
So, 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 – 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.
What features do you lose using a SIP IP phone compared to an

analog phone?
None. In fact, you gain features by using a SIP phone. SIP phones can send
SIP compliant messages hold, transfer, and conference calls.

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 Can sit anywhere on the LAN or WAN.

the server or to a channel
bank.
Flash is used to hold or bring Some IP phones have buttons to do hold, transfer,
up a voice menu to do features conference,…
like conference and transfer.
What features do you lose using a SIP Channel bank (SIP Phone
Gateway) over a T1 or E1 channel bank?
None. SIP channel banks (SIP Phone Gateways) could take a lot less
hardware, depending on your usage numbers. Why? Because a SIP phone only
uses a resource when there is an active call to that phone (rather than tying
up a dedicated T1/E1 channel for each phone like traditional channel banks).
Take a site with 400 business users and assume only 25% of the phones are
in use at any given time. With T1 Channel banks this would take 17 T1s. With
E1 Channel banks this would take 13 E1s. With HDSI this would take 4 HDSI
cards. With SIP channel banks this would use 100 IP resources. Note that
besides figuring out “trunking”, you need to consider “stationing” usage. The
closer the phones have to a 100% usage number (like in a call center), the
less gain you get from SIP channel banks.

What features do you lose using a SIP gateway compared to bringing
analog or digital trunks directly into the Interaction Center?
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.
What type of SIP phones can be used?

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.
Where can these phones physically sit?

Anywhere. The IP phone and the Interaction Center server communicate via
the SIP protocol over IP. Be aware the voice over the network does require
QoS (Quality of Service) configured in your equipment.
What type of integration do you do with SIP phones? What happens
when I hit the hold button on the phone?
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.
What gateways can be used?

A: Any SIP compliant gateway that Interactive Intelligence has certified. This
will allows The Interaction Center to connect to any device on your WAN or
even allows you to have your PSTN connections into the gateway.
Describe your connectivity.

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.
Is a SIP proxy server required?

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.
Does the Interaction Center benefit from or take advantage of a SIP

Proxy Server?

Yes, the Interaction Center can be configured to send its SIP requests to a SIP
proxy (see section 17.2.5 “SIP Proxy Page”).
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?
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 18 “Defining Global Configurations SIP Stations” about
configuring Persistent connections.
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.
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 18 “Defining Global
Configurations SIP Stations” about configuring Persistent connections.

4.4 Implementation Overview Diagrams
4.4.1 Picture: SIP Hardware Approach Overview

The Interaction Center SIP stack uses the SIP protocol to setup and tear
down Voice over IP (VoIP) calls. The audio for SIP calls uses the Real Time
Protocol (RTP). The RTP audio gets put on the internal telephony bus, just
like audio for ISDN calls or audio from analog phone sets. Since all RTP
audio is on the telephony bus, all features such as call analysis (dialer),
conferencing, recording, monitoring, mixing with analog phones, mixing with
trunks (ISDN, E1, T1, Analog) are available with SIP.
SIP can 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.
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
5 When is a SIP Proxy Needed

This is an important question when laying out the topology and cost of your network.
The Interaction Center has special software that sometime alleviates the need for
SIP proxies. First it is important to understand SIP message routing.
5.1 SIP Message Routing

The routing of SIP messages can be done by different devices:
• Some SIP Phones

• Some SIP Gateways
• SIP Proxies
• Interaction Centers
Routing of Inbound SIP messages:

• SIP Proxies receive SIP messages from gateways (or SIP-capable PSTNs)
and route the SIP messages to the Interaction Center or to unmanaged
phones.
• Gateways receive PSTN calls, then convert them to IP, and route SIP
messages to either a primary or backup Interaction Center
• Some gateways are capable of load balancing between a bank of Interaction
Centers
• Interaction Centers can route calls to managed phones
Routing of Outbound SIP messages:

• SIP Proxies receive SIP messages from Interaction Centers or unmanaged
phones and route the SIP messages to gateways (or SIP-capable PSTNs)
• Some gateways are capable of routing SIP messages to WANs, LANs, or the
PSTN
• Some SIP Phones are capable of routing local calls to local gateways,
emergency calls to local gateways, and other calls to either a primary or
backup Interaction Center
• Interaction Centers can route calls to different gateways
LAN
SIP Proxies can be
used to route SIP Interaction
Messages. The SIP M
Phones, Gateways, e s s ag Centers
SIP Phones and Gateways e P at
and Interaction h
Centers can use the
Proxy for all the
routing decisions.
SIP Proxy Server
Some IP Phones LAN

can route SIP Interaction
Messages. If the
specific phone can Centers
not route SIP
messages, then a
SIP Proxy must be SIP Message Path
used. SIP Phone
Some Gateways
can route SIP PSTN / WAN LAN
Messages. If the
specific gateway SIP Gateway Interaction
can not route SIP SIP
M essa Centers
messages, then a ge P
a th
SIP Proxy must be
used.

5.2 Phone Specific Routing
Some hard and soft SIP phones can only do simple addressing (send the SIP
message to a single IP address) while some can do a considerable amount of
routing. If SIP message routing is required and the phone can not do
SIP messaging routing, then a proxy is required, unless the routing
can be done by the Interaction Client. The Interaction Client can send
its telephony requests to either a primary or backup Interaction
Center server.
Type of routing typically needed by a SIP phone:
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.
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.
5.3 When is a Proxy Needed (for the Phone)
See the SIP 3rd Party Component Feature Matrix spreadsheet for the values in the
Decision Tree below.
The network has both a primary and a backup Interaction Center. Is

a SIP proxy required?
A proxy may be required, depending on the capabilities of the SIP phones

and SIP gateways that are used.
First, check if the SIP phones require a proxy. Check the “Backup Proxy”
capability in the SIP 3rd Party Component Feature Matrix spreadsheet.
If “Backup Proxy” is “Yes” or “N/A”, then the phones don’t require a proxy.
If “Backup Proxy” is “No”, then:
• If using the Interaction Client to make calls, no SIP proxy is
needed. Why? Because when the Client makes a call, it sends a
makecall request to the Interaction Center server, which will place
a call to the phone associated with the Interaction Client.

• If dialing from a phone, a proxy server will be required. Since the
phone has no backup proxy capability, the proxy will send the
phone’s outbound call request to the correct Interaction Center
server.
Next, you must check your SIP gateways (if you are using them) to see if
they require a proxy to do similar routing logic.
Can my phones route calls to a local gateway based on what is

dialed (i.e. 911 or 8-555-1234)? If “Yes”, is a SIP proxy required?
Check the “Dial Plan Routing” capability in the SIP 3rd Party Component
Feature Matrix spreadsheet.
If “Dial Plan Routing” is “Yes”, then the answer is:
Yes, the phone can route calls to a local gateway based on what is
dialed. No proxy is needed to do this routing.
If dialing using the Interaction Client, no proxy is needed. The
Interaction Center will have to be configured to send these calls from
that user to that specific gateway.
If dialing using the phone, no SIP proxy is needed. The phone’s
dialplan will do the routing.
If “Dial Plan Routing” is “No”, then the answer is:
The phone can not route calls to a local gateway based on what is
dialed.
If dialing using the Interaction Client, no proxy is needed. The
Interaction Center will have to be configured to send these calls from
that user to that specific gateway.
If dialing using the phone, a SIP proxy is required (since the phone
does not support a dialing plan).
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?
A proxy may be required, depending on the capabilities of the SIP phones
and SIP gateways that are used.
First, check if the SIP phones require a proxy. Check the “Backup Proxy”
capability in the SIP 3rd Party Component Feature Matrix spreadsheet.
If “Backup Proxy” is “Yes”, then the phones don’t require a proxy.
If “Backup Proxy” is “No”, then the phones require a proxy server
(remember, dialing from the Interaction Client is not possible if the
Interaction Center server is unreachable). Since the phone has no backup

proxy capability, the proxy will send the phone’s outbound call request to
the correct Interaction Center server.
Can my phone automatically route calls to a local gateway if both the

primary and backup Interaction Centers can not be reached? If
“Yes”, is a SIP proxy required?
Depends.
If dialing using the Interaction Client, “No”. Since both the primary
and backup Interaction Center are not reachable, the Interaction Client
can not complete a call.
If dialing using the phone, “Yes”, but a SIP proxy is required. The
current phones do not have the ability to have multiple backup proxy
servers (the primary Interaction Center is the main proxy for the
phone, the backup Interaction Center is the backup proxy for the
phone, and the gateway would need to be the second backup proxy for
the phone).
5.4 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 used in the network topology, the 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)
5.5 When is a Proxy Needed (for the Gateway)

The same rules apply for the gateways as for the phones.

6 Connectivity Overview
The following is a bare bones Interaction Center
6.1 Trunk Interfaces with the Interaction Center

Any combination of trunk or station interfaces can be combined on a single
Interaction Center Server.

6.2 Station Interfaces with the Interaction Center
Any combination of trunk or station interfaces can be combined on a single

Interaction Center Server.

7 Connectivity In Detail
7.1 PSTN Connectivity Options

One or all of the options below can be mixed on same system!!!
1. No gateway (tradition 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
1 2 3 4
IC Servers IC Servers IC servers with IC Servers with no
with no with ISDN SIP gateway, using SIP
gateways, connections to connections to connections to
using ISDN gateways gateways PSTN/WAN
connections to
the PSTN
Gateway No gateway. Connect to the Connect to the No gateways necessary.
Features PSTN PSTN and PSTN and PSTN and WAN
connectivity is WAN via WAN via connectivity is done via
done via the tradition traditional SIP. This is not available
telephony connections connections yet, but is coming soon
boards. (ISDN, Frame (ISDN, Frame by large carriers.
Relay) and then Relay) and then
connect to the convert all
IC server via traffic to SIP.
traditional
connections
(ISDN,…).
Are Telephony Tradition ISDN Tradition ISDN Optional. With Optional. With the
boards (or T1, E1, (or T1, E1, the hardware hardware platform
needed? Analog) Analog) platform (telephony boards), IP
telephony telephony (telephony boards are used to do
boards are boards are boards), IP the do the RTP and
used to connect used to connect boards are transcoding. With the
to the PSTN. to the gateway. used to do the software platform (Intel
do the RTP and HMP),
transcoding.
With the
software
platform (Intel
HMP),

1 2 3 4
IC Servers IC Servers IC servers with IC Servers with no
with no with ISDN SIP gateway, using SIP
gateways, connections to connections to connections to
using ISDN gateways gateways PSTN/WAN
connections to
the PSTN
For switchover Yes. The Yes. The No. All No. All connections to
(primary and traditional traditional connections to the IC server are done
backup IC connections connections the IC server via SIP. With SIP, the
servers), is a (such as ISDN) (such as ISDN) are done via switchover routing is
data probe go through the go through the SIP. With SIP, done over the LAN.
needed to route data probe, data probe, the switchover
the digital which routes which routes routing is done
lines? the connections the connections over the LAN.
to the to the
appropriate appropriate
server. server.
N+1 The calls are The calls are The calls are The calls are distributed,
Configuration distributed, by distributed, by distributed, by by the PSTN, across the
(multiple IC the PSTN, the gateways, the gateways, IC servers, simply by
servers) across the IC across the IC across the IC sending the SIP
servers, by servers, by servers, simply messages to different IP
sending the call sending the call by sending the addresses.
to different to different SIP messages
ISDN trunks. ISDN trunks. to different IP
addresses.

7.2 Phone Options
One or all of the options below can be mixed on same system!!!
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 with No. Tradition T1/E1 boards
communicate to the Phone Media Gateway with SIP. The for channel banks, or analog
the phones Phone Media Gateway then station boards are used to
communicates with the phone the same connect to analog stations.
way a traditional channel bank does.
Are resources No. IP resources are No. IP resources are only used when Yes. The phone uses a
used when phone only used when there is there is a voice connection. physical resource even when
is idle? a voice connection. it is idle.
Does the phone No. The SIP hard or No. The Phone Media Gateway is simply Yes. The phone has a
have to be directly SIP soft phones are an IP device anywhere on the network physical connection to the IC
connected to IC simply IP devices (LAN or WAN). server.
server? anywhere on the
network (LAN or WAN).
Phone Types Many vendors make Standard analog phones (2500 sets) and Standard analog phones
supported SIP hard and SIP soft PBX digital phones can be connected to a (2500 sets).
phones. wide variety of Phone Media Gateways.

7.3 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 remote
proxy) 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
dialtone, 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.
7.3.1 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.

7.3.2 Cisco’s SIP SRST (Survivable Remote Site Telephony)
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.

7.3.3 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 WAN
Intelligence’s
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.

7.3.4 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.

7.4 Understanding the Audio Path
7.4.1 Remote Sites Without Remote Gateways

All calls originated from the remote phones are sent to the Interaction Center at the
Central Site.
Advantages: Every call can be recorded and monitored, calling can be done from
the Interaction Client or the phone, every call shows on the Interaction Client.
Disadvantages: None.
7.4.2 Remote Sites with Remote Gateways

Currently, with release IC 2.2, the audio will flow from the phone at the remote site
to the Interaction Center, and then from the Interaction Center to the gateway (or
directly to the telephony card connected to the PSTN). If the gateway is at the
central site, no problem. If the gateway is a telephony card in the Interaction Center
server, no problem. 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.
Advantages: Features, such as recording, monitoring, and conferencing are
all available.
Disadvantages: The audio will be taking two trips across the network, which
will use bandwidth and add delay.
2. IC 2.3 will redirect the audio so the audio stays at the remote site (the audio
is not sent to the central site unless necessary for recording, monitoring, or
conferencing). Also, with a future release, multiple Interaction Centers will
be able to work as one, so an Interaction Center could be added to the
remote site so the audio never leaves the site, even when advanced features
such as recording, monitoring or conferencing are used.
Advantages: The call audio does not take a round trip to the central site.
However, the Interaction Center Server is fully aware of the call. Dialing
can be done from either the phone or the Interaction Client. The audio
can be sent to the central site dynamically if needed (if recording or
monitoring are requested).
Disadvantages: None.
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.
Advantages: The call audio does not take a round trip to the central site.
Disadvantages: The Interaction Client can not 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).

8 Typical Sizing
8.1 IP Resources
Each IP session will use an IP resource. An IP session is either:
• An active SIP connection from a gateway (typically an external call).
• An active SIP connection from the Interaction Center to a managed phone.
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 2 IP
resources.
8.2 Bandwidth Usage

Each IP session will use 2 half duplex connections. Each connection will use
approximately 16 Kbps for 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 (8 for voice, 16 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 wills save bandwidth on silent connections, and not send
silence. Since on a normal conversation there is only one talker and one
listener, using VAD will cut the bandwidth roughtly in half.
So, a G.729 session using VAD will use 24Kbps (24Kbps for the talker and VAD
for the listener).
8.3 Sample Systems

See section 14 “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).
8.4 External Audio Path (in 2.3)

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.
Device A to IC
Direction AudioPath SIP Details

Message
A to IC Internal INVITE Contains A’s advertised codecs.
or
External
IC to A Internal OK IC will use the find the first codec in the codec
or list configured for Device A in IA that matches
External a codec in A’s advertised codecs and return
that codec in the OK.
A to IC Internal ACK Audio can now start.
or
External
Now one IP resource on the IC server is being used. Note that the call might be
destined for device B, it will first come to the Interaction Center. The Interaction
Center will use an IP resource (this allows IVR, dial by name, fax detection, …).

•
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.
Direction AudioPath SIP Details
Message
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:
• For external audio and the checkbox
“Let external devices determine codecs
is selected”, the INVITE contains A’s
advertised codecs and A’s IP address
and port number.
• For external audio and the above
check box is not selected, the INVITE
contains the intersection of A’s
advertised codecs, A’s configured
codecs in IA, and B’s configured
codecs in IA – and A’s IP address and
port number.
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 • For external audio and the checkbox
“Let external devices determine codecs
is selected”, the re-INVITE contains B’s
advertised codecs and B’s IP address
and port number.
• For external audio and the above
check box is not selected, the re-
INVITE contains the original negioated
codec and B’s IP address and port
number.
For internal audio path, two IP resources on the IC server is being used.
For external audio path, zero IP resources on the IC server are being used.
Rules:
• Devices must be SIP (i.e a SIP gateway and an IP phone; or two IP phones).
A ISDN trunck coming into the IC server will always have Must configure
codecs if 2 devices are to talk to each other.
• To insure G.729 is used by remote phone, then must make that the only
codec configured. Otherwise, another codec could be used.

9 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.
9.1 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.
For a Cisco QoS overview, see: http://www.cisco.com/warp/public/732/Tech/qos/
and for a QoS design guide, see:
http://www.cisco.com/warp/public/cc/pd/iosw/ioft/iofwft/prodlit/difse_wp.htm

9.1.1 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.
Vendor specific settings of the Type of Service byte:
• AudioCodes IP boards set this byte, by default, to 0xA0 (=1010 0000), and
can be changed in the line configuration section in Interaction Administrator.
• Intel HMP set this byte, by default, to 0x00, and can be changed in the line
configuration section in Interaction Administrator.
• Cisco IP phones set this byte, by default, to 0xA0 (=1010 0000).
• Messenger 4.6 sets this byte to 0x00.
• Messenger 4.7.0041 sets this byte to 0xA0 (=1010 0000).
9.1.1.1 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 Delay Throughput Reliability Cost Unused

to 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
9.1.1.2 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
9.1.2 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.
9.2 Echo
For a Cisco Echo overview, see
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper091
86a00800d6b68.shtml
See the trouble shooting section for echo (section 35.3 “Echo”).
9.3 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).
CDialogicIPLinkResource::LogRTPQoS(): ipmB1C2 local report: tx packets 1326, tx octets

228072, rx lost packets 0, jitter (hi) N/A, jitter (lo) N/A, jitter (avg) 0
CDialogicIPLinkResource::LogRTPQoS(): ipmB1C2 remote report: tx packets 1245, tx octets

199200, rx lost packets N/A, jitter (hi) N/A, jitter (lo) N/A, jitter (avg) N/A
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.
Transmit Info
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
Receive Info
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.

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

products).
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.
10 VPN, Firewalls, Security, and Network Address Translation
10.1 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 (17.2.3 ”SIP Authentication Page“)
and in the station configuration (19.2.4”SIP Authentication Page”).
10.2 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. By
opening port 5060 (the default port used for SIP) in 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.
10.2.1 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

10.3 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 0 “Creating and Configuring SIP
stations in Interaction Administrator”.
11 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.
12 Inbound Logic / DID

When a SIP call arrives at the Interaction Center, it has a
• destination address (the SIP “To” Header) in the form sip:user@host:port
• origination address (the SIP “From” Header) in the same format.
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 11, “Notes
About User and Station Extensions”).
• When a call is directed to a user, it takes user logic (see section 11, “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 message “To” header 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 CIC 2.2 SR-B/EIC 2.2 GA, tel address schemes were changed to sip
schemes and would never appear. In CIC 2.2 SR-C/EIC 2.2 SR-A, and
2.2 CIC SR-B/2.2 EIC GA hot fixes 1206, 1203, 1212, 1213: For tel
address scheme (addresses that start with “tel:”), the address will look
like tel:number, with no port number.
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_RedirectionTn This is the number that is receiving the redirected call.
Set from the SIP message URI address.

In CIC 2.2 SR-B/EIC 2.2 GA, this attribute is only set if the URI
matches the IP VoiceMail Direct server parameter.
In CIC 2.2 SR-B/EIC 2.2 GA hot fixes 1178 and 1146 and in CIC 2.2
SR-C/EIC 2.2 SR-A, this attribute is set if the Diversion header is
present.
For example, if a phone sip:4003@sip.inin.com:5060 redirects the

call to sip:voicemail@204.180.46.185, Eic_Redirection Tn would be
sip:voicemail@204.180.46.185.
Eic_RedirectingTn This is the number that is redirecting the call.
In CIC 2.2 SR-A/EIC 2.2 GA, this is the exact address in the Diversion
header. Type and port number are NOT added as they are to the
Eic_LocalTn, Eic_RemoteTn, and Eic_RedirectionTn.
SR-C/EIC 2.2 SR-A, 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 CIC 2.2 SR-A, Eic_RedirectingTn is the top most header (i.e. the last
diverted user).
In CIC 2.2 SR-B/EIC 2.2 GA, Eic_RedirectingTn is the bottom most
header (i.e. the first diverted user).
SR-C/EIC 2.2 SR-A, this attribute is set if the Diversion header is

present.
For example, if a phone sip:4003@sip.inin.com:5060 redirects the

call to sip:voicemail@204.180.46.185, Eic_ RedirectingTn would be
sip:4003@sip.inin.com:5060.
Eic_ReasonForCall “U” for Unknown
“B” for Busy
“N” for No answer
“D” for Direct
“A” for Always
In a hot fix and in CIC 2.2 SR-C/EIC 2.2 SR-A, this attribute is set if the
Diversion header is present.
In the above example, Eic_ReasonForCall would be “N”

Eic_ReasonForCallString Exact SIP reason in the Diversion header
In a hot fix and in CIC 2.2 SR-C/EIC 2.2 SR-A, this attribute is set if the
Diversion header is present.
Eic_OutboundSetupParams Requires Hot Fix 2.2 CIC SR-D 1877, 2.2 EIC SR-A HF 1878, 2.2 CIC
SR-E, or 2.2 EIC SR-B.
The value in the call attribute Eic_OutboundSetupParams will be put in
the ININAttr header in the SIP message during outbound call logic and
blind transfer logic.
Note: Eic_OutboundSetupParams can not contain any double quotes.
Eic_OutboundSetupParams syntax:
[name=value[;name=value]*]
When a call is received, call attributes name1, name2,… will be set to

value1, value2,…
Example:
If Eic_OutboundSetupParams is set to:
Agent=Frank Smith;Number In Queue=12
Then the outbound SIP message will include this header:
ININAttr: “Agent=Frank Smith;Number In Queue=12”
And the inbound call will have the follow attributes set before the
incoming call handler is invoked:
Agent will be set to: Frank Smith
Number In Queue will be set to: 12
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 32 “Configuring the Managed Phone Shortcut”), or
• Eic_LocalTn is compared against the IP Message Button server parameter
(section 31 “Configuring Message Waiting Indicators (MWI)”), or
• Eic_RedirectionTn is compared against the IP Voicemail Direct server
parameter (section 30 “Configuring Voice Mail For Non-Managed Phones”).
7. Do IP VoiceMail Direct logic (see section 30 “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.
13 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
(see section 18 “Defining Global Configurations SIP Stations”), an audio connection
might already be established, which means that the request call will start
immediately.
Eic_OutboundSetupParams Requires Hot Fix SR-D XXXX.

The value in the call attribute Eic_OutboundSetupParams will be put in
the ININAttr header in the SIP message during outbound call logic and
blind transfer logic.
Eic_OutboundSetupParams can not contain any double quotes.
The header is the SIP message will appear as:

ININAttr: “value”
Value should have the syntax:

name=value[;name=value]*
When a call is received, the attributes name1, name2,… will be set to

the value1, value2,…
Example:
If Eic=OutboundSetupParams is set to

Agent=Frank Smith;Number In Queue=12
Then the outbound SIP message include this header
ININAttr: “Agent=Frank Smith;Number In Queue=12”
And the inbound call will have the follow attributes set
Agent will be set to: Frank Smith
Number In Queue will be set to: 12

14 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.
14.1 Platform Combinations and Supported Status
Telephony IP-enabled Supported?

Platform Telephony
Boards
Intel/Dialogic None Yes. See section 16 for configuration details.

Software needed.
(HMP) Implemented
in software.
Aculab AudioCodes Yes (this is the preferred hardware configuration). See section
Hardware IP boards 15 for 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 IP boards validated for IC 2.2 for the addition of one (1) AudioCodes card in
Hardware 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
(http://www.inin.com/support/cic/22/hardware/serverlist.asp?q=670&
) and for new installs
(http://www.inin.com/support/cic/22/hardware/download/New_Install
_Server_Matrix.xls)
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 system with PCI IP boards (all IP boards are PCI).
Hardware
14.2 Platform Comparison

AudioCodes IP Boards Intel/Dialogic HMP Software
Model • H.100 [PCI] No telephony boards required. This is a
Numbers IPM260A/120/NoSpan/H100/MVIP/N3 ver. complete software solution.
P03 (ulaw or alaw, 120 RTP sessions). Was
IPM-260A-120 HIP-CI-3 ver. P03.
• H.100 [PCI]
IPM260A/60/NoSpan/H100/MVIP/N3 ver.
• H.100 [PCI]
IPM260A/30/NoSpan/H100/MVIP/N3 ver.
Price Check with your hardware vendor. Check with Interactive Intelligence.
Density 30, 60, and 120 simultaneous RTP sessions. HMP 1.1: See the HMP chapter for the
latest densities (section 16 “Installing and
Benefit: The actual number of usable resources Configuring Intel HMP Software Solution”)
ports provided may exceed the rated 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 the NA. This is a total software solution
boards per new worst case scenario numbers. These 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 1.0: No.
Capable
HMP 1.1: Yes, dual processors.
HMP 2.0: Yes, quad processor and
processor affinity.
Play and No. Must use an additional resource board. I3 does Yes.
Record not use the AudioCodes voice resources and the
board is priced accordingly. Voice resources from
Aculab or Intel/Dialogic cards are used for audio.
• Note: Aculab Prosody boards (each with up to
4 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 does Yes.
not use the AudioCodes conferencing resources.
Usable T1/E1 The versions of these AudioCodes boards do not No. This is a total SIP solution. SIP
Interfaces have network interfaces. gateways must be used if T1/E1 interfaces
are required.
Telephony Bus H.100/PCI
Computer Bus NON-universal PCI. None.
Universal PCI boards coming with 4.2 firmware
(around IC 2.3 timeframe).
Coders G.711, G.723, G.729, GSM, G726. HMP 1.1: G.711, G.723, G.729.
RFC 2833 Yes. This feature is new in CIC 2.2 SR-A/EIC 2.2 Yes
DTMF GA.
VAD Yes No
Echo 30ms of echo cancellation on the voice going from HMP 1.1: No.
Cancellation the TDM bus to the IP network.
Coming in later release.
4.2 Firmware will increase the to 64 or 128ms.
draft-ietf-avt- No. ??
rtp-cn-06
(defines VAD Coming in AudioCodes firmware 4.0 (IC 2.3).
and CNG for
codecs (such
as G.711 and
G.726) that do
not explicitly
define VAD and
CNG).
RTP Starts at 4000 (configurable) and steps up by 10 Starts at 49152 (not configurable) and steps
for the next session. up by 2 for the next session.
Sync RTP is used. Sync RTP is used.
Fax • Clear channel (which is not reliable) • Clear channel (which is not reliable)
• T.38 is supported with CIC 2.2 SR-D. • T.38 planned for CIC 2.2 SR-D HF.
Modems • Clear channel (which is not reliable) • Clear channel (which is not reliable)
Used with Yes No. This is a total software solution. No
Aculab hardware is required.
systems?
Used with Yes No. This is a total software solution. No
Intel/Dialogic hardware is required.
systems?
Vendor 3.91-001-000: CIC 2.2 SR-A and EIC 2.2 GA. HMP 1.1.
Software
3.91-001-004: CIC 2.2 SR-B. Fixes full duplex/half
duplex timing issue with some Cisco switches, add
support for additional 30 and 60 port boards.
3.91-001-009: CIC 2.2 SR-C and EIC 2.2 SR-A.
Fixes max phone problem.
4.00-266-000: CIC 2.2 SR-C HF 1484, EIC 2.2 GA
HF 1504, EIC 2.2 SR-A, and CIC 2.2 SR-D. Fixes
G.711 VAD and static issue.
4.0-384-000: CIC 2.2 SR-C HF 1581, EIC 2.2 SR-
A HF 1582, CIC 2.2 SR-D, EIC 2.2 SR-B.
4.0-389-000: CIC 2.2 SR-C HF 1680, EIC 2.2 SR-
A HF 1674, CIC 2.2 SR-D HF 1673, EIC 2.2 SR-B,

EIC 2.2 SR-E.
4.2: planned for IC 2.3.
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
(future)
15 Installing and Configuring AudioCodes Boards
Check for support status and vendor combination in section 14.1 ”Platform
Combinations and Supported Status” before using AudioCodes IP Boards.
15.1 Important Notes and Restrictions

AudioCodes boards have made newer versions of their boards. This newer
version is Universal PCI (compared to the older non-Universal PCI). The newer,
Universal PCI board will fit into the new servers. The Universal PCI boards will
be supported in 2.3.
AudioCodes boards don't have a special setup like Dialogic and Aculab. 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.
15.1.1 Servers
The certified servers for AudioCodes boards can be found at
http://www.inin.com/support/cic/22/hardware/serverlist.asp?q=670
15.1.2 Known Issues
Issue Workaround Affected Release Fixed Hot Fixes

Releases In
AudioCodes
Multiple gateways are Audiocodes Audiocodes
not supported. A Firmware 3.9 and Firmware 4.2
gateway redirect sent 4.0
back to AudioCodes
will result in one way
audio.
When used with IC 2.2 IC 2.3
Dialogic boards, and
an external SIP call is
started, and the call
external party picks up
immediately, the first
500ms of audio will be
missed. If the external
call rings for over

500ms, no audio will
be missed.
15.1.3 AudioCodes with Dialogic
15.1.3.1 Physical Placement of Boards

If placing AudioCodes boards in a Dialogic system, 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.
15.1.3.2 Dialogic’s 3rd Party Board Configuration

Dialogic’s 3rd Party Device configuration should NOT be done for AudioCodes
boards.
15.1.3.3 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:
o Right click (in the DCM) on “Bus-0”
o Set “Using Secondary Master (User Defined)” to “No”

15.1.4 AudioCodes with Aculab
15.1.4.1 Physical Placement of Boards

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.
15.1.5 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.
15.2 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. The MAC address will be needed for
configuration later in this chapter.

15.3 AudioCodes Switch Port Configuration
WARNING: 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.
Two options:
1. Always set the switch port configuration to auto-negotiate if the switch is
capable of full-duplex. See your switch vender for its configuration capabilities.
2. Set the server parameter “AudioCodes Port Duplex” (see section 34 “Server
Parameters”) to match the switch setting.
The negotiated speed and duplex are logged in the event viewer.
15.4 AudioCodes Plug and Play Drivers (wdpnp.sys, ipm260.inf)

When the AudioCodes hardware is first installed and the system is powered up,
the operating system will bring up the hardware wizard to help with
configuration of the new boards drivers. The AudioCodes plug and play drivers
are copied onto the server by newer Interaction Center installs (earlier versions
of the Interaction Center will need to manually download and install the
appropriate AudioCodes plug and play driver hotfix from the Interaction
Intelligence support site).
Interaction Center Product Release/Hotfix for AudioCodes Plug and Play drivers
CIC 2.2 SR-D Install HF 1713 to install the PnP driver (5.0.5.1).
OLD: PnP driver 5.0.5.0 is installed with SR-D.
Must activate PnP drivers below to prevent a resource conflict.
CIC 2.2 SR-C Install HF 1713 to install the PnP driver (5.0.5.1).
OLD: Install HF 1549 to install the PnP driver (5.0.5.0).
CIC 2.2 SR-B Install HF 1713 to install the PnP driver (5.0.5.1).
EIC 2.2 SR-A Install HF 1714 to install the PnP driver (5.0.5.1).
If the PnP drivers are not activated, you could have a TsServer startup problem
caused by a resource conflict. The TsServer log’s last line is an attempt to
open the AudioCodes board:

CIPLinkResourceMgr::OpenBoardsAndChannels(): AudioCodes
board 0, set to slave (0)
The resource conflict can be solved by activating the PnP drivers with the
hardware wizard.
If the Hardware Wizard fails, try this manual steps:

1. Get the wdpnp.sys and ipm260.inf files into the correct directories (newer installs do
this automatically, older installs you must do this manually – see table above).
Steps 2, 3 and 4are only needed if you have manually disabled the boards via the
device manager.
2. On the IC server, bring up the Device Manager and remove the device associated
with the AudioCodes card. Usually, it’s listed as a System DMA Controller or
Unknown PCI card. If the plug-n-play drivers have been previously installed then it’ll
show up as a device under the Jungo container. Do not remove any device that
they are not sure is the AudioCodes device.
3. Power down the IC server and remove the AudioCodes card.
4. Power up the server and verify that the device removed in step 2 is gone in the
Device Manager. Verify that there aren’t any !’s (bangs) on any devices.
5. Power down the server and install the AudioCodes card.
6. Power up the server, when the device manager detects the AudioCodes card, follow
the procedures below.
7. Verify there aren’t any !’s on the PCI devices in the Device Manager dialog.
8. Start IC.
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. Select Next to select

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

15.5 Installing the AudioCodes PCI Driver (windrvr.sys)
AudioCodes PCI driver is copied to your local hard drive when you install the
Interaction Center software with Dialogic or Aculab telephony support.
However, you must register the driver manually. To do this, complete the
following steps. Note: You must be logged on as administrator or a user with
comparable privileges to complete these steps.
1. Verify the driver was installed with the Interaction Center install (this
occurs automatically). The driver file should be located at
C:\winnt\system32\drivers\windrvr.sys
2. To insure the driver is previously added, you must remove the driver.
This is done by clicking Start -> Run… and entering “cmd” to bring up a
DOS prompt. Then run the program
“D:\I3\IC\Server\Diagnostics\AudioCodes\wdreg_gui remove”. If the
default path for the Interaction Center install was not used then replace
the “D:\I3” with the install path that was used.
Ignore any error message wdreg_gui remove displays since the drivers
might (or might not) were already added.
3. Add the driver to the list of devices that the operating system loads on
boot. This is done by clicking Start -> Run… and entering “cmd” to
bring up a DOS prompt. Then run the program
“D:\I3\IC\Server\Diagnostics\AudioCodes\wdreg_gui install”. If the
default path for the Interaction Center install was not used then replace
the “D:\I3” with the install path that was used. The following dialog will
appear if the drivers are successfully installed.
Note that if you run “wdreg_gui install” twice, you will get a failure
message. To fix, simply run “wdreg_gui remove” and then “wdreg_gui
install” again.
4. Reboot the computer to start the PCI driver.

5. Verify the PCI driver was appropriately installed by running the
“C:\I3\IC\Server\Diagnostics\AudioCodes\GUI_3.90.05.exe”
application. If the AudioCodes board is installed and the driver was
successfully installed then the aforementioned application will output
the AudioCodes boards detected on the local PCI bus as shown in the
dialog below. Do not use this application to configure the boards.
This will be done via Interaction Administrator later in this
chapter.

15.6 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. This involves two
steps, adding AudioCodes to the Vendor Support server parameter and adding
the AudioCodes specific server parameters listed below.
Adding the Vendor Support server parameter
1. Start Interaction Administrator
2. Add AudioCodes to the Vendor Support server parameter. To do this,
locate the Vendor Support server parameter, double-click on it and add
“;AudioCodes” to the end of the current Parameter Value. Click OK to apply
the changes.
Adding the AudioCodes Audio Adjustment server parameters
New in CIC 2.2 SR-C HF 1462 and EIC 2.2 GA HF 1463: There are two
AudioCodes specific server parameters for audio adjustment.
These server parameters been replaced with the gain settings in the line,
station, and global station in Interaction Administrator and will only work if
those values in IA have not been changed; If the values in IA have been
changed, the values in IA will be used. The IA parameters were added in CIC
2.2 SR-D.
These server parameters are retired in 2.3.
Note that these configurations are dynamic, and will take affect when the next
SIP call starts.
o AudioCodes Network Gain (optional, default is 0, valid range: –31 to 31
(dB)). 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 an agent’s phone.
o AudioCodes Bus Gain (optional, default is 0, valid range: –31 to 31 (dB)).
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.
Adding the AudioCodes specific server parameters

There are four AudioCodes specific server parameters that can be set in the
server parameters section of Interaction Administrator:
• AudioCodes Setup (required at all sites)
• AudioCodes Firmware Path (required at all sites)
• AudioCodes Law Select (optional, default is ulaw)
• AudioCodes Start Media Port (optional, default is 4000)

AudioCodes Setup
This server parameter is used to configure the network settings for each
AudioCodes board in the system. It has the following syntax,
Board1Configuration;Board2Configuration;…;BoardNConfiguration
where each board configuration (BoardNConfiguration) contains,

MACAddress;MasterOrSlave;BoardIP;SubnetMask;DefaultGateway
Each BoardNConfiguration has the following form,

MACAddress;MasterOrSlave;H100Termination;BoardIP;SubnetMask;DefaultGateway
An additional parameter (H100Termination) was added in the following

releases:
2.2 CIC SR-A HotFix 1019 (Aculab) and HotFix 1031 (Dialogic)
2.2 CIC SR-B
2.2 EIC GA
Omit this parameter if running on an earlier.
Example 1: 00908f12ab89;0;0;10.1.3.50;255.255.255.0;10.1.3.1
Board with MAC address 00908f12ab89 is configured as a
bus slave and does not terminate the H.100 bus. It is
assigned IP address 10.1.3.50, subnet mask 255.255.255.0,
gateway 10.1.3.1.
Example 2: 0;0;0;172.16.128.76;255.255.0.0;172.16.1.1
First board discovered is configured as a bus slave and does
not terminate the H.100 bus. It is assigned IP address
172.16.128.76, subnet mask 255.255.0.0, gateway
172.16.1.1.
Example 3: 00908f12ab89;0;1;172.16.128.76;255.255.0.0;172.16.1.1
Board with MAC address 00908f12ab89 is configured as a
bus slave and terminates the H.100 bus. It is assigned IP
address 10.1.3.50, subnet mask 255.255.255.0, gateway
10.1.3.1.
MACAddress 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. Note: If the board is not required to
have a specific IP address then 0 (zero) can be entered in
this field and the system will assign the IP address to the
next discovered board that does not have a specific IP
assigned to it.
MasterOrSlave 1 or 0, represents 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
set to zero (slave).
H100Termination H.100 termination is new with 2.2 CIC SR-B and 2.2 EIC GA.
Omit this parameter on earlier systems (and put your boards
as if 0 were selected). 1 or 0, represents whether the board
should terminate the H.100 bus or not, respectively. If the
board is situated on is the first or last board on the bus then
hardware termination should be set to one. Otherwise, if the
card is between Dialogic or Aculab cards then H.100
termination should be set to zero.
BoardIP IP address assigned to the board. Enter it in dotted decimal
format, for example, 10.12.1.15. WARNING: This IP
address must not be 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.
SubnetMask Subnet mask of the network to which the AudioCodes board
will be attached. Enter it in dotted decimal format, for
example, 255.255.0.0
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).
AudioCodes Firmware Path

This server parameter 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.
AudioCodes Law Select

This parameter changes the encoding scheme of the TDM bus. The default
type is uLaw. Set this server parameter to change the default type. Valid
values are aLaw and uLaw.
AudioCodes Start Media Port

This server parameter changes the starting port for AudioCodes RTP sessions.
The default value is 4000. If this port conflicts with other resources or
applications then set this server 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, 4001 and 4002 for RTP, RTCP, and T38 fax,
respectively. The next IP resource will consume 4010, 4011, 4012 and so on.
Before CIC 2.2 SR-D and EIC 2.2 SR-B, if the number entered was not
divisible by 10, then the AudioCodes firmware would fail to start.

Restarting the Interaction Center
The Interaction Center needs to be restarted after adding AudioCodes to the
Vendor Support server parameter and adding the AudioCodes Setup server
parameter to configure the boards.

16 Installing and Configuring Intel HMP Software Solution
Use this section if using Intel HMP (Host Media Processing). This is a total
software solution.
16.1 Important Notes and Restrictions

HMP 1.0 is no longer supported. HMP 1.1 is the supported release.
HMP Overview: http://www.intel.com/network/csp/pdf/7786wp.htm
16.1.1 Servers
The certified servers for HMP can be found at
http://www.inin.com/support/cic/22/hardware/serverlist.asp?q=670
16.1.2 Densities
These limits apply to HMP 1.1.
HMP Resource HMP 1.1 Limitations Notes

Total number of Total <= 254 The sum of all resources.
resources R+V+F+C/2 <= 254
R+V+C <= 254
R RTP G.711 (G.711 R<= 120 The number of RTP resources for any
IP Resources) R>= max (V,C,F) given configuration should be greater
than or equal to the number of voice,
conferencing, or fax resources
(whichever requires the highest
number of resources).
E Enhanced RTP E<= 64 This is the number of G.711
(Adding E<= R resources (above row) that are
G.723/G.729 to the capable of low bit rate coding.
G.711 IP
Resources)
V Voice Resources V<= 120
V<= R
S Speech Recognition S<= 120 Usually 0 (unless echo cancellation
Resources S<= V is needed for speech rec). This is the
number of voice resources (above
row) that will have CSP (continuous
speech processing) capabilities (used
for echo cancellation with ASR).
Note that these might not be needed
in an IP environment, depending on
the echo cancellation of the gateways
and devices.
C Conference C<= 120
Resources C<= R
F T.38 Fax Resources F<= 32 Note that T.38 is not available yet
F<= R (plus planned in the near future).
IP Call Control NA Always 0. This is not needed since
Interactive Intelligence’s SIP stack is
used for the call control

16.2 Vendor Software
The Intel/Dialogic HMP software:
• Install Intel HMP Release 1.1 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.
16.3 Configuring your HMP system.

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

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
section 17.1 “Line Configurations not exposed through Interaction
Administrator”), a Windows 2000 registry setting from
http://support.microsoft.com/default.aspx?scid=KB;en-us;248611& must be
set.
16.3.2 IP addresses
Currently, 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.
16.3.3 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.
By default, the APIC timer is used.
Problem 1: The local APIC is disabled and unavailable for use.
Solution: The HMP system will detect the presence and state of the local APIC timer by
performing software checks at initialization. If an operational APIC is detected, it will be used. If
an APIC is not detected, the HMP system will default to using the mmtimer.
Problem 2: 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
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
Change the value of
HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\DlgcApicDrv\Start from 2 to 4 (to
disable the service). 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
Using Microsoft KB article #Q237556 (http://support.microsoft.com/default.aspx?scid=KB;en-

us;q237556), change the computer type to “Standard PC”.
16.4 Known IC Issues
Issue Workaround Affected Releases Release Fixed In
HMP reports as a None – no bad side IC 2.3

unsupported release. affects except the log
message.
Futures
T.38 Coming in a future IC EIC 2.2 SR-A 2.2 SR or hot fix.

release.
CIC 2.2 SR-D
Enhanced RTP resources Coming in a future IC EIC 2.2 SR-A 2.2 SR or hot fix.
(low bit rate codecs) are not release.
managed correctly if less CIC 2.2 SR-D
than the number of G.711
IP resources.
Speech recognition is not Coming in a future IC EIC 2.2 SR-A IC 2.3
supported with IC on HMP. release.
CIC 2.2 SR-D
16.5 Known HMP Issues
Issue Workaround Affected Releases Release Fixed In
Major Issues
No major issues at this

time.

Minor Issues
HMP does not support Restart Intel service if HMP 1.1

semi-automatic mode in the restarting the IC service.
service. Without it, IC can
not automatically restart the
Intel service if IC is
restarted.
HMP service does not start This is a problem with all
automatically when set to Dialogic releases. Always
automatic with the Services set the service mode from
applet (works when you set the DCM OR go to
to automatically from the HKEY_LOCAL_MACHINE
DCM). \ SYSTEM \
CurrentControlSet \
Services. For Dialogic
dependonservice key, add
“dlgcmcd”
Hardware Futures
HMP 1.1 supports a single None. HMP 1.1

processor (without hyper-
threading), a single
processor (with hyper-
threading), or dual
processors (with no hyper-
threading).
Software Futures
AGC is not supported None. HMP 1.1

(Dialogic hardware IP
boards supported AGC on
the packet transmit side).
No Echo cancellation. None. HMP 1.1
No Transaction Record. Without Transaction Planned for 2.0.
Record, conference
resources are required to
do supervisory recording.
Receive and Transmit Gain HMP 1.1 Planned is to implement ALC
adjustment on individual (Automatic Level Control) on
streams is not supported the inbound IP side.
(Dialogic hardware IP
boards supported non-
dynamic gain adjustment on
the board, not individual
streams, and only on the
packet receive side).
If a RTP stream is no longer
received, it is not reported
as not be received.
If a firewall provides an odd This may well be a Cisco
port number (via PAT) for PIX issue – it is unclear
RTP, HMP will still transmit from the RTP RFC.
on the even port. Investigating….
Starting port (49152) is not
configurable.

17 Creating and Modifying SIP Lines in Interaction Administrator
SIP lines allow the IC Server to communicate with the SIP boards.
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.

17.1 Line Configurations not exposed through Interaction Administrator
Interaction Center Values Description

Parameter (use
DsEditU in the line
configuration to
set
“Disable Delayed • “No” (default) If “Yes”, delayed media negotiation will be
Media” • “Yes” disabled and the codecs will be advertised in the
INVITE message.
Note that delayed media negioation will not
require any IP resources to ring a group of
phones.
Note that early media from gateways can not be
received unless using normal media (i.e. can not
use early media).
Requires HF CIC 2.2 SR-C Hot Fix 1562, EIC 2.2
SR-A Hot Fix 1564, CIC 2.2 SR-D, or EIC 2.2 SR-
B.
"RTP QOS Byte" • 0xA0 (default) QOS byte.

• any 2 byte hex Important: If using dseditu, the field should be
number. entered as “0xA0”.
For HMP platforms, a Windows 2000 registry
setting must also be set. See
http://support.microsoft.com/default.aspx?scid=
KB;en-us;248611& for more information.
Requires HF CIC 2.2 SR-D Hot Fix 1638, EIC 2.2
SR-A Hot Fix 1670, CIC 2.2 SR-E, or EIC 2.2 SR-
B.
“DTMF Payload” • 101 (default) The value used for the DTMF RTP payload type.
• 96 - 127 100, 102-105 should not be used for
AudioCodes.
This value is also in the station and the global
station configuration.
B.
Modify • “|3600” Registration Interval in seconds.
"RegistrarIPList" (default) Requires HF CIC 2.2 SR-D Hot Fix 1638, EIC 2.2
• “|number” SR-A Hot Fix 1670, CIC 2.2 SR-E, or EIC 2.2 SR-
B.
“Inband Transfer • “No” (default) IC will use the 302 or the REFER (depending on
Enabled” • “Yes” whether the call has been answered or not) to do
a blind transfer on an external SIP call.
SR-A Hot Fix 2000, CIC 2.2 SR-E.
17.2 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
from the drop-down lists, 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.
17.2.1 SIP Configuration Page

The SIP Configuration Page is a multi-tabbed dialog. This section offers
reference information on configured this dialog.

General
Configuration
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.
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.
In CIC 2.2 SR-C/EIC 2.2 SR-A, or with the appropriate

hot fix, the domain name is also used to select an IP
address to be used in SIP messages generated by the
Interaction Center (IC) when the IC contains multiple
Network Interface Cards (NICs). For example, if an IC
had two NICs, one for a private subnet (192.168.1.100)
and one for the company LAN (172.16.5.20), the domain
name could be used to select the company LANs IP
address to be used in all SIP messages. The domain
name is resolved into an IP address and IC chooses a NIC
address that matches the first byte of the domain IP
address. For example, the domain name might be
company.com, and company.com resolves (through DNS)
to 172.16.5.1. In this case, the IC would choose the
company LAN IP address of 172.16.5.20.
Silence Time Same as in today’s Line objects
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
Auto Disconnect when Same as in today’s Line objects. Linked to Silence Time.
Silence is Detected Default: Off
Voice Activate Checked (On) means use VAD on any connection that is
Detection (VAD) 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.

General
Configuration
Terminate Analyses On Checked (On) means, if call analyses is used, to terminate
Connect 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
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) support RFC2833.
AudioCodes hardware boards support RFC2833.
Disable T.38 Faxing Unchecked (Off) means that T.38 will be used for faxes
over SIP.
Default: Off.
Vendor Specific
Intel/Dialogic software (HMP) does not support T.38
(planned for HMP 1.1).
AudioCodes hardware boards support T.38.

General
Configuration
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 30ms of echo
cancellation on the voice going from the TDM bus to the IP
network.
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) does not support network
gain.
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) does not support bus gain.
AudioCodes hardware boards support bus gain.
Maximum Number of
Calls
Combined If the Combined radio prompt is selected, the Combined value
Inbound/Outbound prompt is shown.
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
Outbound that the SIP line will process. When the maximum is reached, no
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.

Maximum Number of
Calls
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.
17.2.2 SIP Protocol Page
Protocol
Transport Protocol Can be UDP or TCP. If UDP is selected, the following additional
fields must be defined. Most of the following fields are grayed
out if TCP is selected.
Default: UDP
Note: TCP is not supported in this release.

Receive Port UDP/TCP: Port number for which the IC SIP engine will be
servicing requests.
Valid: 1024 to 65535
Default: 5060
T1 Timer UDP: Timer value in milliseconds that represents the initial
incremental delay between packet retransmission.
Valid: 500 to T2 (milliseconds)
Default: 500
T2 Timer UDP: Timer value in milliseconds that represents the maximum
incremental delay between packet retransmissions.
Valid: 4000 plus (milliseconds)
Default: 4000
Maximum Packet Retry UDP: Maximum Packet Retry for requests
Valid: 0 to 10
Default: 10

Protocol
Maximum Invite Retry UDP: Maximum packet retry for INVITE and ACK requests
Valid: 0 to 6
Default: 6
17.2.3 SIP Authentication Page
Authentication
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.

17.2.4 SIP Compression Page
New in CIC 2.2 SR-D, CIC 2.2 SR-C Hot Fix 1372 and EIC 2.2 SR-A:
Delay media negotiation is simply delaying the advertising of supported codecs in the
SIP codec negotiation process. Delayed media on outbound calls gives the
Interaction Center server more control over the coder negotiation process.
If one codec is used, normal media negotiation is used for outbound calls. If more
than one coder is selected, delayed media negotiation is used for outbound calls.
The calling system always controls what time of media timing occurs.
Delayed media negotiation can also be disabled by setting “Disable Delayed Media”
setting “Disable Delayed Media” (see section 17.1 “Line Configurations not exposed
through Interaction Administrator”) – “Disable Delayed Media” requires HF CIC 2.2
SR-C Hot Fix 1562, EIC 2.2 SR-A Hot Fix 1564, CIC 2.2 SR-D, or EIC 2.2 SR-B.

Compression
Codecs 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.
Notes:
• Note 1: Only configure the codecs the platform supports.
• Note 2: These codecs are not supported by Intel/Dialogic Software
HMP 1.1. Only configure the coders the platforms support.
• Note 3: HMP does not support 4 frames/packet with G.723 and does
not support 1 frame/packet with G.729.
• Note 4: 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 5: 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 6: 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.

Name Data Default Possible Default Possible Compression MOS
Rate Frame Frame Sizes Frames Frames Delay
Size /Packet /Packet
Note 4 Note 5 Note 6

G.711 mu-law 64Kbps 20ms 10,20,30ms 1 1 0.75 ms 4.1
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 3
G.729AB 8Kbps 20ms 20ms 2 1,2,3,4 10 ms 3.7
Note 3
GSM 061.0 13.2Kbps 20ms 20ms 1 1,2,3,4
Note 2
G.726 32Kbps 20ms 20ms 1 1 1 ms 3.85
Note 2

17.2.5 SIP Proxy Page
The following page is primarily used to specify a priority list of proxy
addresses for an outbound proxy.
Proxy
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

17.2.6 Registrar Page
This page is used to configure SIP registrar information required by IC.
Registrar
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

18 Defining Global Configurations SIP Stations in Interaction
Administrator
18.1 Global Station Configurations not exposed through Interaction

Administrator

Parameter (use
DsEditU in the
global
configuration to
set)
“Allow SIP • “No” (default) Allow the station’s contact information be
Registration” • “Yes” automatically set from Contact header in the
station’s INVITE or REGISTER message.
This value is also in the station configuration.
B.
AudioCodes.
This value is also in the station and the line
configuration.
B.
"Connection Call • 5 (default) The value (in seconds) that a non-persistent
Warm Down Time" • 0 – no connection will be disconnected if not reused.
warmdown Requires HF CIC 2.2 SR-D Hot Fix 1843, EIC 2.2
time, SR-A Hot Fix 1844, CIC 2.2 SR-E, or EIC 2.2 SR-
disconnect B.
connection
when idle
IC supports a default configuration for all the SIP station (note that each station has
the ability to overwrite one or all of these configurations). To get to this
configuration, go to Interaction Administrator, select (on the left pane) the container
that is your server name. “Configuration” will be displayed on the right pane. Double
click on “Configuration” and then select the “SIP Station” tab. The following dialog
will be shown.
18.1.1 Notes on “Allow SIP Regitration” and the audio-enabled client.

The Interaction Center client (when running in audio-enabled mode) will use the
Microsoft RFC 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
18.2 Global Station Configuration Dialog

Global SIP Note that all these values are used by default by each station. If
Station Configuration: SIP desired, each station has the ability to individually configure
Station these options.
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. If none is specified, the
line group selection is done through tradition dial plan dialing.
If you have many lines on a system, the system could perform
better if the line group is specified.
Use Proxy for Station Checked indicates that the proxy list configured in the line
Connections configuration in Interaction Administrator should be used to
connect stations. Unchecked the Interaction Center will contact
the stations directly.
Voice Activate Detection (VAD) Checked indicates to use VAD when connecting to this station.
Uncheck indicates that VAD will not be used. This value
overwrites the VAD value selected in the line.
Default: Off
Vendor Specific
Intel/Dialogic software (HMP) does not support VAD.
Station Connections are Checked indicates that connections to the station are persistent,
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.

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
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.

DTMF Type The type of DTMF signaling. 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) support RFC2833.
AudioCodes hardware boards support RFC2833.
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 30ms of echo cancellation
on the voice going from the TDM bus to the IP network.
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 an agent’s phone.
Vendor Specific
Intel/Dialogic software (HMP) does not support network gain.
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) does not support bus gain.
AudioCodes hardware boards support bus gain.

Global SIP Note that all these values are used by default by each station. If
Station Configuration: desired, each station has the ability to individually configure
Authentication these options.
Station Authentication Use this dialog to enter authentication information for all SIP
station. If desired, each station has the ability to individually
configure these options. Enabling authentication forces the
phone to exchange credentials 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.

19 Creating and Configuring SIP stations in Interaction
Administrator
19.1 Station Configurations not exposed through Interaction

Administrator

Parameter (use
DsEditU in the
station
configuration to
set)
“Allow SIP • “No” (default) Allow the station’s contact information be
Registration” • “Yes” automatically set from the Contact header in the
station’s INVITE or REGISTER message.
B.
AudioCodes.
This value is also in the line and the global
station configuration.
B.
"Connection Call • 5 (default) The value (in seconds) that a non-persistent
Warm Down Time" • 0 - infinite connection will be disconnected if not reused.
B.
19.2 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).
See section 33, “Sample Configurations” for an overview.

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 18 “Defining Global Configurations 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.
19.2.1 General Page

The Station Configuration dialog appears as shown in the following figure after
selecting SIP in the Connection box.

Enter the extension and complete the following options:
SIP
Station Configuration
Active Check indicates this configuration is to be used.
Use Default If checked, the values configured in the Global SIP Station
Configuration (see section 18 “Defining Global Configurations
SIP Stations”).
• Use Proxy for Station If the “Use Default” check box is checked, the values
Connections configured in the “Global SIP Station Configuration” will be
• Voice Activate Detection (VAD) used. If the check box is not checked, then these values will
• Station Connections are be able to be set independently. See the Global SIP Station
Persistent Configuration (see section 18 “Defining Global Configurations
• Terminate Analysis On Connect SIP Stations”) for complete description.
• Number of Call Appearances
per Station
• DTMF Type
• Echo Cancelation
• Network Gain
• Bus Gain
Connection This is the SIP address of the SIP device. This address is
used by the Interaction Center to connect to this SIP station.
See below for details.
Identification This is the SIP address that identifies the SIP device. This
address is used by the Interaction Center to identify this SIP
station. See below for details.
19.2.2 Connection SIP Address Page

You can now have the contact address of the stations be dynamic. See setting
“Allow SIP Registration” in the global station configuration (see section 18.1
”Global Station Configurations not exposed through Interaction Administrator”)
and station configurations (see section 19.1 ”Station Configurations not exposed
through Interaction Administrator”). Requires CIC 2.2 SR-D HF 1638, EIC 2.2 SR-
A HF 1670, CIC 2.2 SR-E, or EIC 2.2 SR-B.
When you press the “Connection” button, the following dialog appears:

Connection This is the SIP address of the SIP device. This address is used by the Interaction
User 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 7111@1.1.1.1. 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 (see section 17.2.1 “SIP Configuration Page”).
19.2.3 Identification SIP Address Page

When you press the “Identification” button, the following dialog appears. There
are 3 options described below.
Option 1
Option 2 (RECOMMENDED)
Recommended
Option 3

Identification This is the SIP address that is used to identify calls made from the SIP station.
User 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 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.
In the above dialog options 1 and 3, the Interaction Center Server will use the
address 7111@2.2.2.2 to identify the phone. For option 2, the Interaction Center
Server will use the address 7111.
Option 1 (“Same as connection”) should be used if the following is true:

• The phone’s connection address is the same address as its identification
address (the address it sends in the “From” header of the SIP messages).
This can be used for Pingtel phones but not for Cisco phones. Cisco phones
do not put its own address in the “From” header.
Option 2 (RECOMMENDED) (new 2.2 CIC SR-B/2.2 EIC GA, “Use a alternate
format”) must be used if for the phone if all the following is true:
• The phone’s connection address is NOT the same address it sends in the
“From” header of the SIP messages
• Switchover is used
• The extension value configured on the phones is unique amongst all your IP
phones.
Option 3 (“Use a predefined format”) should be used rarely. All the following
must be true for Option 3 to be used.
• The phone connection address is not the same address it sends in the “From”
header of the SIP messages. For Cisco phones, change the host portion to
match the value configured for proxy1_address in the Cisco phone
configuration, which is the addresses of the Interaction Center.
• Switchover is not used.
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 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. For more
info on VPNs, see 10.3 “VPN”.
Continued on next page….

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 3)
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 1).
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).
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 (see section 17.2.1 “SIP Configuration Page”).

19.2.4 SIP Authentication Page
Authentication
Authentication If the “Use Default” 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 values will
be able to be set independently. See the Global SIP Station
Configuration (see section 18 “Defining Global Configurations
SIP Stations”) for complete description.

20 Dial Plan Basics for SIP
20.1 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).

20.2 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.

21 Gateway Configuration
There are three ways to get a call to a gateway:
1. Explicit: 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. Problem with Explicit:
this is cumbersome to dial and forces the dialer to pick the gateway.
2. Proxy: In the proxy section of the line configuration, put the gateway as
the proxy. All calls will be sent to the proxy (which is the gateway).
Problem with Proxy: Assuming no number translation is done in the
dialplan, a dialed number, such as 5551212, will be sent in the SIP
messages as tel:5551212. Most gateways do not handle the “tel:”
scheme on SIP messages. New in 2.3: In 2.3, if “Use Tel” is not set in
the line configuration, the IC server will not use the tel scheme, but will
translate the number to sip:5551212@proxy. This allows no dial plan
“@gateway” translations to be added in the dial plan.
3. DialPlan: In the dial plan (described in this sections below), configure a
translation from 5551212 to sip:5551212@gateway. This works well
with many gateways.

21.1 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

Dial The number to be dialed for the specified input pattern. In the above dialog, the number to be
String 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 is no alpha
characters 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 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 The number to be dialed for the specified input pattern. In the above dialog, the number to be
String 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).

21.2 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.
21.2.1 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.

21.2.2 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.

21.3 Multiple Gateway Configuration
When using multiple gateways, there are two concerns:
1. Detecting gateway failure and/or congestion.
2. Choosing the proper gateway to do the outbound call, and being able to
‘roll’ to another gateway with failure or congestion is detected.
21.3.1 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.
21.3.2 Choosing the Proper Gateway: 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.

DialPlan
Configure the Interaction Center Dial Plan to translate the dialed number to
sip:number@gateway. This is described earlier in this chapter.

22 Fax 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 encapsulates the T.30
data and handling 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.
22.1 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.
22.2 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.
22.3 Scenarios
22.3.1 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.
• The gateway can reject the re-invite; if this occurs 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.
22.3.2 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.
22.4 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.
22.5 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.

22.5.1 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, "fax protocol t38" should be added to "voice service voip":
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
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122ne
wft/122t/122t11/faxapp/t38.htm

23 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.

24 Tie Line and Multi-site Configuration
Benefits: Tie-Lines
Interaction Center A Interaction Center B
WAN
Interaction Center C
LAN LAN
9 Instant connectivity, each SIP IC can connect to other SIP IC’s - virtual tie-lines
9 Great for IC’s running in a Multi-site configuration
9 Direct dialing, 101@A can be dialed, with no configuration changes on either
system, from B or C.
9 ‘Transparent’ dialing between ssstems with Milti-site or with dial plan
configuration
©2003 Interactive Intelligence Inc.
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.

25 Switchover Configuration
Read the Switchover white paper for the most up-to-date switchover information.
25.1 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, a registry

setting must be done (in a future release, this will be done by the Install).
How:
Use regedt32 to go to the following key:
HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\Interaction
Center\ProcessTree\Level1\SwitchoverService.
With the SwitchoverService key selected in the left pane, double click on the
CommandLineArguments entry in the right pane. When the String Editor
dialog box appears, type /NoDataProbe, then click OK.
25.2 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 19 “Creating
and Configuring SIP stations in Interaction Administrator”.
25.3 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.
26 Interaction Client Configuration
26.1 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.

26.2 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.

26.2.1 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:
1. don't run Messenger when running the audio enable Interaction Client.
2. OR run Messenger configured NOT to use a communication service. See
the “SIP 3rd Party Component Application Note” for details.
3. 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.
26.2.2 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.
26.3 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.
To add the Line Details page in Interaction Client:
1. 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.
2. In the Pages dialog, select Line Details form the Available list, as shown in the
following figure.

27 Phone Services
This feature is new in CIC 2.2 SR-C and EIC 2.2 SR-A.
Restrictions:
• The only phones currently supported (by default) are Cisco. However, this
same mechanism can be used by other phones, but is not included in 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, ….
The default features are:
• Log in (using the Interaction Center user extension & password)

• Log out
• Once logged in: call control of current call, change user status (Available,
Out of Office,…), logout
• call control includes: Hold, Transfer, Voice Mail, Record, Alert Supervisor
(notification via email).
All these actions (login, logout, status change, call control) are implemented in
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.
a. Open Internet Information Services

b. Right-click on “Default Web Site” and select “new” and then select “Virtual Directory…”
c. Click Next
d. Enter name of virtual directory

e. Enter Path to installed directory (defaulted in install to c:\i3webs)
f. Take defaults on Access Permissions (need Read and Run scripts)

g. Select “Finish”.
3. Using Interaction Administrator, create or select an account on the IC Server

which will be used by the Phone Services web application to connect to the IC
Server. This user does not require any special permissions. Assign a
password to the account. The default account expected by the installed web
application is CiscoWebClient with a password of CiscoWebClient.
4. On the Web Server, modify i3webs\globals.inc and change the following
application parameters:
• Notifier: This should be the computer name of the Primary IC Server.
• NotifierUserName: This should be a valid IC account selected in step 3
(defaults to CiscoWebClient)
• NotifierPassword: This should be the password assigned to the IC account
selected in step 3 (defaults to CiscoWebClient)
5. On the TFTP Server (where the Cisco phone files are stored), modify the
SIPDefault.cnf file and change the following values:
• services_url: This should be set to the URL of the services start page. If
you want only the I3 Phone services available, use the path to splash.asp
(e.g. http://<WebServer>/i3webs/splash.asp). If you want to include and
configure external web services available for the Cisco Phone, use the path
to servicemenu.asp (e.g. http://<WebServer>/i3webs/servicemenu.asp).
• proxy_backup: In a switchover environment, this should be set to the IP
address of the backup IC server.
6. You must reset Cisco IP Phone in order to get changes to the SIPDefault.cnf
file to take affect on the phone.
7. In a switchover environment, if the web server is not on an IC Server, the web
application will need permissions to add values to the
HKEY_LOCAL_MACHINE\Software\Interactive Intelligence\EIC\Notifier\Servers
registry key. With regedt32 (not regedit), create the registry key and add
permissions for the Internet Guest Account (normally IUSR_<WebServer>) to
have full control.

28 IP Resource Management
The Interaction Center will manage IP resources in a shared pool. Each IP board
that is put into the system, has a fixed number of IP resources. Each IP resource
is capable of carrying an audio connection (via RTP) between a SIP device and the
Interaction Center. These IP resources are put into a shared pool to be used by
connections to SIP phones and gateways. The line configuration is the vehicle to
manage IP resources (see section 17 “Creating and Modifying SIP Lines").
Notes:
• Each SIP line that is created is capable of handling multiple calls. The
maximum number of outbound and inbound calls is configurable.
• Each SIP line must have a unique port number. This port is used to receive
SIP messages.
• To pick which SIP line will be used when the Interaction Center connects to a
SIP stations (i.e. managed stations), see the Line Group configuration in
section 18 “Defining Global Configurations SIP Stations”.
• To pick which SIP line will be used when a SIP station (i.e. managed station)
connects to the Interaction Center, is a matter of port numbers. If the SIP
station is sending its requests to port 5060 (the SIP default), then the SIP
line that has 5060 selected for its port number will be used for the
connections.
Example:
Assume there are 240 IP resources in your system.
Assume there are 500 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 500 SIP
phones will only use up to 125 IP resources at any given time.
The other 115 IP resources (240 minus 125) can be used for inbound and
outbound calls to a SIP gateway. Out of the 115, we want to reserve 100 for
inbound calls.
First, configure for the gateway.

Create “SIP Line-Gateway” (see section 17 “Creating and Modifying SIP Lines”)
with the following values:
• Port: 5060
• Maximum outbound calls: 15
• Maximum inbound calls: 100
Next, configure for the managed phones. Remember, typically you only need
one SIP line. However, with this approach, you are sending part of your SIP calls
on one line (i.e. port 5060) and the other SIP calls on the second line (i.e. port
5061).
Create “SIP Line-Phones” (see section 17 “Creating and Modifying SIP Lines”)
with the following values:

• Port: 5061. You must make sure each phone is directed to send its SP
requests to this port.
• Maximum combined calls: 125
Configure the system so calls to managed stations will use “SIP Line-Phones”:
• Create a line group “SIP LineGroup-Phones” and put the line “SIP Line-
Phones” in it.
• Configure the “SIP LineGroup-Phones” to be used when the Interaction
Center connects to SIP phones (see the Line Group configuration in
section 18 “Defining Global Configurations SIP Stations”).

29 Configuring the Message Button For Voicemail Retrieval
Set the server parameters IP Message Button and Force Message Button Login
according to section 34 “Server Parameters”.
The IP 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 IP Message Button to “9999” will allow IP phones to configure
their message buttons to 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
Interaction Administrator 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.
Vendor Specific
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 9999@172.16.132.16 where 172.16.132.16 is the Interaction
Center’s IP address and 9999 is the value set in the server parameter IP Message
Button.
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 9999@172.16.132.16 where 172.16.132.16
is the Interaction Center’s IP address and 9999 is the value set in the server
parameter IP Message Button.

30 Configuring Voice Mail For Non-Managed Phones (Diversion)
What is a managed phone? What is an unmanaged phone? See section Error!
Reference source not found. “Error! Reference source not found.”.
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.
30.1 Logic
The diverted SIP message will have:
• URI: sip:voicemail@204.180.46.185
• diversion header(s) original destination, and diversion header(s) divert
reason: CC-Diversion: <sip:5858652@siptest.wcom.com>;reason=no-answer
INVITE sip:voicemail@204.180.46.185 SIP/2.0

t: <sip:5858652@siptest.wcom.com>
f: "Fred Flintstone"<sip:cic@i3worldcom.com>;tag=26680
v: SIP/2.0/UDP 204.180.46.185;received=204.180.46.185
i: 5df33e3731a72d7309a756b4571016d2@204.180.46.185
CSeq: 1 INVITE
CC-Diversion: <sip:5858652@siptest.wcom.com>;reason=no-answer
Notes:
• CC_Diversion header is 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
In CIC 2.2 SR-B/EIC 2.2 GA, tel address schemes were changed
to sip schemes and would never appear. In CIC 2.2 SR-C/EIC
2.2 SR-A, and 2.2 CIC SR-B/2.2 EIC GA hot fixes 1206, 1203,
1212, 1213: For tel address scheme (addresses that start with
“tel:”), the address will look like tel:number, with no port
number.
In CIC 2.2 SR-B/EIC 2.2 GA hot fixes 1178 and 1146 and in CIC
2.2 SR-C/EIC 2.2 SR-A, this attribute is set if the Diversion
header is present.
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 CIC 2.2 SR-A/EIC 2.2 GA, this is the exact address in the

Diversion header. Type and port number are NOT added as they
are to the Eic_LocalTn, Eic_RemoteTn, and Eic_RedirectionTn.
2.2 SR-C/EIC 2.2 SR-A, 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 CIC 2.2 SR-B/EIC 2.2 GA, tel address schemes were changed
to sip schemes and would never appear. In CIC 2.2 SR-C/EIC
2.2 SR-A, and 2.2 CIC SR-B/2.2 EIC GA hot fixes 1206, 1203,
1212, 1213: For tel address scheme (addresses that start with
“tel:”), the address will look like tel:number, with no port
number.
In CIC 2.2 SR-A, Eic_RedirectingTn is the top most header (i.e.

the last diverted user).
In CIC 2.2 SR-B/EIC 2.2 GA, Eic_RedirectingTn is the bottom
most header (i.e. the first diverted user).
2.2 SR-C/EIC 2.2 SR-A, this attribute is set if the Diversion
header is present.
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 a hot fix and in CIC 2.2 SR-C/EIC 2.2 SR-A, this attribute is
set if the Diversion header is present.
In the above example, Eic_ReasonForCall would be “N”

Eic_ReasonForCallString Exact SIP reason in the Diversion header
In a hot fix and in CIC 2.2 SR-C/EIC 2.2 SR-A, this attribute is
set if the Diversion header is present.
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 IP VoiceMail Direct server 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 that has the Eic_RedirectingTn (or the user portion) configured as the
user’s extension (in Hot fix for CIC 2.2 SR-C, in EIC 2.2 SR-A and CIC 2.2 SR-
D) or Attribute 2 in the user configuration.

30.2 Setup
Set the server parameter IP VoiceMail Direct according to section 34 “Server
Parameters”.
The IP VoiceMail Direct value should be either the whole SIP address with type and
port number (sip:user@host:port) or just the user portion (user).
Your phones or proxies must be configured to send the calls to the number
configured as the IP VoiceMail Direct number.
Configure, in Attribute 2 in the user configuration in Interaction Administrator, the
address in the diversion header.
Before CIC 2.2 SR-B/EIC 2.2 GA, the attribute 2 value must match exactly the SIP
address received in the diversion header.
In CIC 2.2 SR-B/EIC 2.2 GA hot fixes 1178 and 1146, the attribute 2 value should
EITHER be the user portion (user) OR the whole SIP address with type and port
number (sip:user@host:port) . The address received in the diversion header will get
“sip:” and port number appended (if necessary) to it before the compare occurs.
In CIC 2.2 SR-B/EIC 2.2 GA hot fixes 1206, 1203, 1212, 1213 and CIC 2.2 SR-C/EIC
2.2 SR-A, the attribute 2 value should be:
For sip scheme (addresses that start with “sip:”), attribute 2 should be EITHER the
user portion (user) OR the whole SIP address with type and port number
For tel scheme (addresses that start with “tel:”), attribute 2 should be EITHER the
number portion (number) OR the whole SIP address with type (tel:number). There
is no user or port number with tel addresses.
Make sure you publish the new handlers that are hot fixes in 2.2 SR releases.

Vendor Specific
Cisco: The Cisco phones do not forward calls to voicemail systems. This
responsibilities is done by proxies.
Pingtel: The Pingtel phones can forward calls to voicemail systems. See the “SIP 3rd
Party Component Application Note” for details.
31 Configuring Message Waiting Indicators (MWI)

Set the server parameters Message Light and Message Light Persistent according to
section 34 “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.

For station-less users (i.e. users using Unmanaged Stations) that still require MWI,
attribute 3 in the user configuration of IA can be used (you must have CIC 2.2 SR-
B/EIC 2.2 GA or later for the attribute 3 feature)
• 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). Note that in a future IC release, this address
will be in a different field and will allow a “:” in the address.
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 attribute 3 will be used. This logic is in System_MessageLight.ihd and can be
changed on a site by site basis.

Vendor Specific
Cisco: The Cisco phones do not subscribe for notifications. The Interaction Center
will send unsubscribed notifications to the Cisco phones.
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.
32 Configuring the Managed Phone Shortcut

Set the server parameter IP Managed Phone Shortcut according to section 34
“Server Parameters”.
The IP Managed Phone Shortcut 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 IP 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).

33 Sample Configurations
33.1 Central Site Only, Primary Interaction Center Only, Cisco IP Phones
Specifications
Central Site Yes
Is there a Remote Site? No

Is there a Gateway at the remote site for local and 911 calling? No
Number of Interaction Center Servers (load balancing) 1
Is Switchover being utilized (primary and backup Interaction Center servers)? No
Central Site Proxy Optional

The proxy is optional since the Cisco phones are capable of routing. Other
supported phones might or might not have the same capabilities.
Interaction Center
• Primary Interaction Center’s IP Address: 1.1.1.1
In Interaction Administrator, create a station (see section 0, “Creating and
Configuring SIP stations in Interaction Administrator”)
• Station Name: Station7101

• Station Type: Workstation
• Station Extension: 7101
• Station SIP Connection Address: sip:7101@2.2.2.2:5060 (must match
info from the phone below).
• Station SIP Identification Address: sip:7101@1.1.1.1:5060 (must match
info from the phone below). If using multiple proxies, the value should be
7101 (you must have CIC 2.2 SR-B/EIC 2.2 GA or later for this non-whole
SIP address to work, and your Cisco line1_name parameters must be
unique as recommended in the SIP 3rd Party Component Application
Note).
In Interaction Administrator, create a user
• User Name: User1
• User Default Workstation: Station7101
• User Extension: 101
Interaction Administrator, Phone Container, DID/DNIS
• User DID: 715-7101 (this number is mapped to User1)

Cisco IP Phone
• IP Address: 2.2.2.2
• line1_name: 7101
• outbound_proxy:
If using a proxy, set to the IP address of the proxy.

If not using a proxy, leave empty.
• proxy1_address: Set to the IP address of the Interaction Center (1.1.1.1).
• proxy_backup:
If using a single proxy, leave empty.
If using two proxies, set to the backup.
Interaction Client
The computer that is running the client should be named the same as the
station (“Station7101) OR use the /w=Station7101 (Station Name) flag (see
section 26.1, “Associating the Interaction Client with a Station)
33.2 Central Site Only, Primary and Backup Interaction Centers, Cisco IP
Phones
Specifications
Central Site Yes
Is there a Remote Site? No

Is Switchover being utilized (primary and backup Interaction Center servers)? Yes

The proxy is optional since the Cisco phones are capable of backup routing (in
this case, the phones would use backup routing to the backup Interaction Center
if the primary Interaction Center was not reachable). Other supported phones
might or might not have the same capabilities.
Interaction Center
• Backup Interaction Center’s IP Address: 9.9.9.9

• Station SIP Identification Address: 7101 (you must have CIC 2.2 SR-
B/EIC 2.2 GA or later for this non-whole SIP address to work, and your
Cisco line1_name parameters must be unique as recommended in the SIP
3rd Party Component Application Note).


Cisco IP Phone
• outbound_proxy:
• proxy1_address: Set to the IP address of the Primary Interaction Center
(1.1.1.1).
• proxy_backup:
If not using a proxy, set to the IP address of the Backup Interaction
Center (9.9.9.9).
Interaction Client
Proxy Server
• Optional in this configuration if phones are capable of routing
• Would be configured to send SIP calls to the Interaction Center backup if
the Interaction Center primary is not reachable.
• Could be used to route calls to the remote site’s gateways for long
distance savings.
33.3 Central and Remote Site (no remote gateways), Primary Interaction
Center Only, Cisco IP Phones
Specifications
Central Site Yes
Is there a Remote Site? Yes


this case, the phones would use backup routing only if there were two Cisco
proxies and would use backup routing to the backup proxy if the primary proxy
was not reachable). Other supported phones might or might not have the same
capabilities.
Interaction Center
Configuring SIP stations in Interaction Administrator”). In this configuration,
the stations are the same at the Central and Remote sites.


Cisco IP Phone
• outbound_proxy:
(1.1.1.1).
• proxy_backup:
Interaction Client

Proxy Server
distance savings.
33.4 Central and Remote Site (with remote gateways), Primary Interaction
Center Only, Cisco IP Phones
Specifications
Central Site Yes

Is there a Gateway at the remote site for local and 911 calling? Yes

this case, the phones at the remote sites would use backup routing to their local
gateways if the Primary Interaction Center was not reachable). Other supported
phones might or might not have the same capabilities.
Interaction Center



Cisco IP Phone
• outbound_proxy:
(1.1.1.1).
• proxy_backup:
If not using a proxy, set to the gateway at the remote site.
• Dial plan
If number dialed is local, route call to gateway at remote site.
If number dial is emergency (911), route call to gateway at remote
site.
Interaction Client
Proxy Server
distance savings.
33.5 Central and Remote Site (with remote gateways), Primary and
Backup Interaction Center Only, Cisco IP Phones
Specifications
Central Site Yes

Is there a Gateway at the remote site for local and 911 calling? Yes
Is Switchover being utilized (primary and backup Interaction Center servers)? Yes
Central Site Proxy Required

The proxy is required in this case since the Cisco phone is not capable of backup

routing to 2 different locations (in this case it would need 2 backups: the
gateway at the remote site and the backup interaction center server). Other
supported phones might or might not have the same capabilities.
Interaction Center
• Backup Interaction Center’s IP Address: 9.9.9.9

B/EIC 2.2 GA or later, and your Cisco line1_name parameters must be
unique as recommended in the SIP 3rd Party Component Application
Note).

Cisco IP Phone
• outbound_proxy:
Set to the IP address of the proxy.
(1.1.1.1).
• proxy_backup:
• Dial plan
If number dialed is local, route call to gateway at remote site.
If number dial is emergency (911), route call to gateway at remote
site.

Interaction Client
Proxy Server
• Would be configured to send SIP calls to the Interaction Center backup if
the Interaction Center primary is not reachable.
distance savings.
33.6 Cisco IP phone, no Interaction Client (stand alone lobby phone)

Interaction Center
In Interaction Administrator, create a station (see section 0, “
Note that all these values are used by default by each station. If
Global SIP desired, each station has the ability to individually configure
Station these options.
Configuration:
Authentication
Station Authentication Use this dialog to enter authentication information for all SIP
station. If desired, each station has the ability to individually
configure these options. Enabling authentication forces the
phone to exchange credentials 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.

Creating and Configuring SIP stations”)

• Station Type: Stand Alone Phone
• Station SIP Connection Address: 7101@2.2.2.2 (must match info from the
phone below).
• Station SIP Identification Address: 7101@1.1.1.1 (must match info from
the phone below).
• User DID: 715-7101 (this number is mapped to Station7101)

Cisco IP Phone
• outbound_proxy:
• proxy1_address: Set to the IP address of the Interaction Center (1.1.1.1).
33.7 Microsoft Messenger Soft IP Phone, Interaction Client, User, and

Station
Interaction Center

• Station SIP Connection Address: 7101@2.2.2.2 (must match info from
the phone below).
the phone below).
Interaction Administrator, User Container

Microsoft Messenger
• IP address: 2.2.2.2
• Server name or IP address:
If not using a proxy, set to the IP address of the Interaction Center.
• Sign-In name: 7101@1.1.1.1
Interaction Client
33.8 Microsoft Messenger Soft IP Phone, Interaction Client with Audio,

User, and Station
Interaction Center

• Station SIP Connection Address: 7101@2.2.2.2 (must match info from
the phone below).
the phone below).
Interaction Administrator, User Container

Microsoft Messenger
• No configuration is required
Interaction Client
The computer that is running the client should be named the same as
the station (“Station7101) OR use the /w=Station7101 (Station Name)
flag (see section 26.1, “Associating the Interaction Client with a
Station)

34 Server Parameters
You can set the following values in the Server Parameters container in Interaction
Administrator.

Server Parameter
“IP Managed Phone • none (default) Used as a convenient way for managed phones
Shortcut” • any character to connect to the main IVR.
available from
telephone This value should be either the whole SIP
keypad. address with type and port number
(sip:user@host:port) or just the user portion
(user).
Note: This number is typically a “*”. If managed

phones 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.
“IP Message Button” • none (default) Used for voice mail retrieval over the IP phone
• any number when the message button on the SIP phone is
pressed.
• “No” (default) This value should be either the whole SIP

“Force Message • “Yes” address with type and port number
Button Login” (sip:user@host:port) or just the user portion
(user).
Note: You must configure voicemail button of the

phone to call this number when it is pressed.
Use to indicate whether the user id and password

will be required at all times. By default, the user
id and password are only required if the
Interaction Client is not at an available status.
“IP Voicemail Direct” • none (default) Used to send calls directly to voicemail for
• any number unmanaged phones. Voicemail for managed
phones is already handled.
This value should be match the whole SIP

address in the diversion header exactly, or just
match the user portion.
Note: You must configure your network to send

calls, destined for voicemail, to this number.
“SIP Default Display any string, Used as the SIP display string in the FROM
String” defaults to header when calls are made to persistent SIP
“Interaction managed stations and to any SIP managed
Center” station when the client MakeCall button is
pressed. This string will show on the From field
on the phone display.
“Message Light” If these TRUE activates message light logic in the
parameters are Interaction Center.
present, then
regardless of TRUE will cause the message light on the phone

their value, to persistent in the on state while any unread
they are TRUE. voicemails exist.
“Message Light If they are not FALSE will cause the message light on the phone
Persistent” present, then to turn off after the first unread voicemail is
that indicates a read.
value of
FALSE.
“Ringback File” • none (default) This is the wav file that is played while a call on a
• any wav file SIP line is made. If no wav file is configured,
name the file Ringback.wav in the Resources directory
is used.
“UseOffHookEventFo • No (default) New in TS CIC 2.2 SR-C HF 1599, EIC 2.2 SR-A
rSIPDialing” • Yes HF 1637, CIC 2.2 SR-D HF 1638, EIC 2.2 SR-B
and CIC 2.2 SR-E. Need handler HF CIC 2.2 SR-
C 1601 and CIC 2.2 SR-D 1699 for new offhook
logic. Must republish System_OffHook handler
(new parameter in initiator) and the other
handlers in 1601/1699 if setting this parameter
to Yes.
“AculabConferenceIn 1..27 with 20 New in TS CIC 2.2 SR-C HF 1608, EIC 2.2 SR-A
putVolume” being the default HF 1637, CIC 2.2 SR-D HF 1638, EIC 2.2 SR-B
and CIC 2.2 SR-E. This will be the value of
conference inputs to Aculab conferences.
“TsDiag” “AcuConfDisAgcDi Adding AcuConfDisAgcDiag to the TsDiag server
ag” parameter will turn off AGC in Aculab
conferences.
“AudioCodes Port • Auto (default) New in TS CIC 2.2 SR-D HF 1638, EIC 2.2 SR-A
Duplex” • Half HF 1670, EIC 2.2 SR-B and CIC 2.2 SR-E.
• Full 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 duke
“Broken RTP • 30 (in seconds, New in TS CIC 2.2 SR-D HF 1638, EIC 2.2 SR-A
Disconnect Time” default) HF 1670, EIC 2.2 SR-B and CIC 2.2 SR-E.
• 1..3600 (in In seconds. If no RTP, RTCP, and no comport
seconds) noise packet (used with VAD) is received in the
• 0 or negative configured time, the call will be automatically
disables this disconnected.
feature (NOT Note: If a remote device does, for the time
recommended) 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.
Warning: Setting this field to 0 or negative could
cause no audio behavior since persistent
connections will appear operational to the IC
server while the phone has no connections (it
rebooted).
Vendor Specific
Intel/Dialogic software (HMP) does not support
this parameter.
AudioCodes hardware boards support this
parameter.
New in TS CIC 2.2 SR-D HF XXXX, EIC 2.2 SR-A
HF XXXX, EIC 2.2 SR-B and CIC 2.2 SR-E.
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.
“AudioCodes • 40 (in ms, In milliseconds. Minimum jitter buffer delay that

Minimum Jitter default) will be used by the dynamic jitter buffer
Buffer Delay” • 0..150 (in ms) algorithm on AudioCodes boards. The algorithm
will never reduce the jitter buffer below this
value.
“AudioCodes Jitter 0..12 (7 is the Jitter buffer optimization factor is a unit-less

Buffer Opt Factor” default) 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.
Vendor Specific
Intel/Dialogic software (HMP) does not support
these parameters.
AudioCodes hardware boards support these
parameters.
35 Troubleshooting
35.1 Tracing
The flowing are trace topics for the possible different issues. Use
traceconfig.exe to set these topics to 61, which is one of the Notes levels.
• TsServer | TsServer – Turn on for any TS related investigation.
Turn on the following depending on this problem:
• 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.

35.2 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.
• Insure that all the IP cards are configured with unique IP addresses. If they
have duplicate addresses, then one way or no audio will occur (depending on
the ARP protocol). Special thanks to Jason P. for configuring his topology in
this unique manner.
• Insure that the subnet mask is correct.
• If using Intel HMP 1.0, make sure you are on a single processor machine.
• If using Intel HMP, make sure the IP address of your network card is the same
that is in the registry (see section 16.3.2 “IP addresses”).
• If using Audiocodes do these strings appear in the TsServer log: msg='RTP

Dest IP Address Un Reachable” and msg='Activate RTP/RTCP command
failure”. This implies the cable is not plugged into the AudioCodes board or
there is a network issue.
• Are you running Microsoft Messenger (or another SIP stack) on the
Interaction Center Server? If so, by default, both multiple SIP stacks will
conflict with each other.
• Using Microsoft messenger and it is requesting an odd port number?
35.3 Echo
• Understand echo (see section 9.2 “Echo”).
• Note the direction of the echo. Are the agents hearing their own voice or are the remote
callers hearing the echo?
• Use the RTP Audio Monitor and Analysis Guide to record the audio directly from the
network (see section 36.5 “RTP Audio Monitor and Analysis Guide”)
• Adjust the gain parameters. For EIC 2.2 SR-A and CIC 2.2 SR-C this is done with
server parameters AudioCodes Network Gain and AudioCodes Bus Gain. For CIC 2.2
SR-D, these values are in Interaction Administrator line, station, and global station
configuration dialog boxes.
1. If the phones are transmitting too “hot”, the agent could hear their own voice. In
this case, the Network Gain can be slowly turned down to help overcome the
phone’s transmit levels.
• Some head phones generate echo.
• Some phones generate echo if their volume is turned too high.
35.4 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 49 “Voice Issues on Networks ”).
• Check the RTP Sender reports in the TsServer log (see section 9.3 ”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.
35.5 DTMF Problems
35.5.1 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. If having DTMF problems, the options are:
• Try using RFC2833. This is the best solution. For configuring DTMF type
to RFC2833, see sections 17.2.1 “SIP Configuration Page” for lines and
19”Creating and Configuring SIP stations in Interaction Administrator”.
• Try using G.711.
• If using Intel HMP, make sure the IP address of your network card is the same
that is in the registry (see section 16.3.2 “IP addresses”).
35.5.2 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.
35.5.3 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.
35.6 Miscellaneous
35.6.1 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.
35.6.2 All incoming calls going immediately to held state

• Is the IP address of the IP board set to 0.0.0.0?
35.6.3 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 0
“Creating and Configuring SIP stations in Interaction Administrator”).
35.6.4 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 0 “Creating and Configuring SIP stations in Interaction
Administrator”).
35.6.5 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.
35.6.6 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.
35.6.7 Managed station not ringing

• Check the SIP Contact Address of the station in Interaction Administrator (see
section 0 “Creating and Configuring SIP stations in Interaction Administrator”).
• 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.
35.6.8 Message Button playing the main menu

• Check the number that is configured on the SIP device. It should match the
number configured in section 29 “Configuring the Message Button For Voicemail
Retrieval”.
35.6.9 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 26.2 “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.
35.6.10 “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
26.2.1 “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.
35.6.11 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.

35.6.12 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.
36 Tools
36.1 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
36.2 Coder Bandwidth Usage

http://www.voip-calculator.com/calculator/lipb/
36.3 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:
o Select UPD (on the left side)
o Select Throughput (on the right side)
o Load Qcheck on two servers (one server should be the Interaction Center server).
o 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).
o Swap the servers in the drop down list and get the results for audio in the other
direction.
36.4 Speakeasy
Test upload and download speeds: http://chi.speakeasy.net/
36.5 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?

37 Index
5.1 Layer 3 Type of Service Byte, 50, 51 IVR, 21

ACD, 21 Lines, 81
AudioCodes boards, 60 MWI, 139
installing, 63 Outbound Logic, 58
AudioCodes Firmware server parameter, 74 Overview, 17
AudioCodes Law Select server parameter, Diagrams, 28
74 With traditional interfaces (no IP),
AudioCodes Setup server parameter, 73 35
AudioCodes Start Media Port server Proxy configuration, 92, 101, 102, 103, 107
parameter, 74 Registrar configuration, 93
Boards Server parameters, 152
list of supported, 13, 28, 38, 60 SIP
Compatibility, 21 Documentation, 17
Compression configuration, 87, 88, 89 Global configuration, 94
DID, 55 Lines, 81
Documentation, 17 Overview, 17
Extenstions Station configuration, 100
assigning, 55 SIP Phones
FAQ, 21 Comparing to Analog, 24, 25
Gateways, 26 Handling more than two calls, 27
HMP (Host Media Processing) Stations, 100
Installing and Configuring, 13, 61, associating with Interaction Client,
76 123, 124, 125
Inbound Logic, 55 SUP Phones
Installing
Supported phones, 26
AudioCodes, 63 Terms, 17
Interaction Client Troubleshooting, 154
configuration, 125 Type of Service, 77
IP Managed Phone Shortcut server Voice Mail
parameter, 141 configuration for non-managed
IP Resource Management, 129, 133
phones, 136
IPLink Boards
Voice Mail Button
configuring in Interaction
configuring, 135
Administrator, 77, 81 Voice over IP
installing, 76 Notes, 47, 49, 50
type of service configuration, 77 Voicemail, 21
IPLInk Boards, 60

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SIP Application Note

Installation and Reference Guide

Interaction Center 2.2

Last updated 10/20/2004

Always check for a newer version of this document!

7601 Interactive Way

Interactive Intelligence, Inc.

Interaction Center Platform Statement

How do I know if I have a documented feature?

SIP Application Note 2 of 159 © 2004 Interactive Intelligence, Inc.

2 Where can I get information? ............................................................... 12

5 When is a SIP Proxy Needed ................................................................ 27

6 Connectivity Overview .......................................................................... 32

SIP Application Note 3 of 159 © 2004 Interactive Intelligence, Inc.

7 Connectivity In Detail ........................................................................... 37

8 Typical Sizing ....................................................................................... 46

9 Voice Issues on Networks .................................................................... 49

10 VPN, Firewalls, Security, and Network Address Translation............... 52

11 Notes About User and Station Extensions .......................................... 54

12 Inbound Logic / DID .......................................................................... 54

SIP Application Note 4 of 159 © 2004 Interactive Intelligence, Inc.

16 Installing and Configuring Intel HMP Software Solution .................... 75

17 Creating and Modifying SIP Lines in Interaction Administrator ......... 80

18 Defining Global Configurations SIP Stations in Interaction

SIP Application Note 5 of 159 © 2004 Interactive Intelligence, Inc.

19 Creating and Configuring SIP stations in Interaction Administrator .. 99

20 Dial Plan Basics for SIP.................................................................... 107

21 Gateway Configuration .................................................................... 111

22 Fax Configuration ............................................................................ 118

23 Modem Configuration....................................................................... 121

SIP Application Note 6 of 159 © 2004 Interactive Intelligence, Inc.

25 Switchover Configuration ................................................................ 123

26 Interaction Client Configuration ...................................................... 124

27 Phone Services ................................................................................ 128

28 IP Resource Management ................................................................ 132

29 Configuring the Message Button For Voicemail Retrieval ................. 134

30 Configuring Voice Mail For Non-Managed Phones (Diversion).......... 135

31 Configuring Message Waiting Indicators (MWI) .............................. 138

32 Configuring the Managed Phone Shortcut ........................................ 140

33 Sample Configurations ..................................................................... 141

34 Server Parameters ........................................................................... 151

SIP Application Note 7 of 159 © 2004 Interactive Intelligence, Inc.

36 Tools ................................................................................................ 158

37 Index ............................................................................................... 160

SIP Application Note 8 of 159 © 2004 Interactive Intelligence, Inc.

Typo corrections 1/16/2002

Updated IPLink firmware names. 1/21/2002

More cautions, such as leading 0s in IP address 1/25/2002

More info on configuring call appearances 2/28/2002

Add AudioCodes setup information. 3/1/2002

Add info on SIP addresses ¾/2002

Hardware restrictions 3/21/2002

More Q&A charts, Outbound logic section 3/26/2002

VPN, Gateway selection 4/24/2002

Power Usage, better description of id field 5/28/2002

SIP Application Note 9 of 159 © 2004 Interactive Intelligence, Inc.

Edited content for typos. No substantive content changes. (SMS) 7/22/2002

Diversion header info, when is a proxy needed (chapter 5) 8/13/2002

Dialogic PTR bundle 1, Audiocodes card placement in Dialogic system, phone

SIP Application Note 10 of 159 © 2004 Interactive Intelligence, Inc.

New illustrations in Chapter 6: Connectivity Overview

Combined all gateway selection into a single chapter

Warmdown time of 0 is wrong, addinged “Inband Transfer Enabled” server