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Always check for a newer version of this document!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

AbstractThis 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 Information1994 2004 Interactive Intelligence Inc. All rights reserved. Interactive Intelligence, Interaction Center Platform, Communit, 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, Mobilit, Virtual Office powered by the Enterprise Interaction Center, Vocalit, 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 StatementThis 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) Communit

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 grayedout. 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.


2 of 159 2004 Interactive Intelligence, Inc.

Table of Contents1 2 Change Log............................................................................................. 9 Where can I get information? ............................................................... 12 2.1 2.2 2.3 2.4 2.5 Interactive Intelligence Web Site.................................................... 12 Third Party Component Certification ............................................... 12 Software Versions and Upgrades .................................................... 12 Whats New..................................................................................... 12 Known Issues with Interaction Center Products ............................. 14 Hot Fixes .....................................................................................14 Known Issues...............................................................................14

2.5.1 2.5.2 2.6 3 4

Known Issues with Other Products ................................................. 14

Glossary of Terms................................................................................. 16 Introduction ......................................................................................... 16 4.1 4.2 Available SIP-Related Application Notes ......................................... 16 Standards........................................................................................ 17 Other Companies..........................................................................17 What is an RFC.............................................................................17 SIP Standards ..............................................................................18 Why has RFC 2543 been replaced with RFC 3261?.............................18 IP Address and Ports .....................................................................19 Security Alert ...............................................................................19

4.2.1 4.2.2 4.2.3 4.2.4 4.2.5 4.2.6 4.3 4.4

SIP Q&A .......................................................................................... 20 Implementation Overview Diagrams............................................... 27 Picture: SIP Hardware Approach Overview .......................................27

4.4.1 5

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

6

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



6.2 7

Station Interfaces with the Interaction Center ............................... 33

Connectivity In Detail ........................................................................... 37 7.1 7.2 7.3 PSTN Connectivity Options.............................................................. 37 Phone Options................................................................................. 40 Remote Survivability and Emergency Dialing .................................. 41 Ciscos NON-SIP SRST (Survivable Remote Site Telephony) ................41 Ciscos SIP SRST (Survivable Remote Site Telephony)........................42 Interactive Intelligences Remote Survivability using SIP ....................43 Emergency (911) Dialing using SIP .................................................44 Remote Sites Without Remote Gateways..........................................45 Remote Sites with Remote Gateways...............................................45

7.3.1 7.3.2 7.3.3 7.3.4 7.4 7.4.1 7.4.2 8

Understanding the Audio Path ........................................................ 45

Typical Sizing ....................................................................................... 46 8.1 8.2 8.3 8.4 IP Resources ................................................................................... 46 Bandwidth Usage ............................................................................ 46 Sample Systems .............................................................................. 46 External Audio Path (in 2.3)............................................................ 47

9

Voice Issues on Networks .................................................................... 49 9.1 Quality of Service (QoS).................................................................. 49 Layer 3 IP Header Byte..................................................................50 Layer 2 Byte (802.1p/Q)................................................................51 9.1.1 9.1.2 9.2 9.3

Echo ................................................................................................ 51 RTCP Sender Reports ...................................................................... 51 VPN, Firewalls, Security, and Network Address Translation............... 52 Security........................................................................................ 52 Firewalls and NAT ........................................................................ 53 Cisco Firewall Information ...........................................................53 VPN .............................................................................................. 54

10

10.1 10.2 10.3 11 12 13 14

10.2.1

Notes About User and Station Extensions .......................................... 54 Inbound Logic / DID .......................................................................... 54 Outbound Logic.................................................................................. 57 Platforms ........................................................................................... 59 Platform Combinations and Supported Status .............................. 59 Platform Comparison.................................................................... 59

14.1 14.2



15

Installing and Configuring AudioCodes Boards .................................. 62 Important Notes and Restrictions ................................................ 62 Servers ....................................................................................62 Known Issues ............................................................................62 AudioCodes with Dialogic ............................................................63 AudioCodes with Aculab ..............................................................64 a-law and mu-law ......................................................................64 15.1.1 15.1.2 15.1.3 15.1.4 15.1.5

15.1

15.2 15.3 15.4 15.5 15.6 16

Prerequisites ................................................................................ 64 AudioCodes Switch Port Configuration ......................................... 65 AudioCodes Plug and Play Drivers (wdpnp.sys, ipm260.inf) ........ 65 Installing the AudioCodes PCI Driver (windrvr.sys) ..................... 69 Configuring the AudioCodes Boards with Interaction Administrator 71

Installing and Configuring Intel HMP Software Solution .................... 75 Important Notes and Restrictions ................................................ 75 Servers ....................................................................................75 Densities ..................................................................................75 16.1.1 16.1.2

16.1

16.2 16.3

Vendor Software .......................................................................... 76 Configuring your HMP system. ..................................................... 76 QoS Setting ..............................................................................76 IP addresses .............................................................................76 Timers .....................................................................................76

16.3.1 16.3.2 16.3.3 16.4 16.5 17

Known IC Issues .......................................................................... 77 Known HMP Issues....................................................................... 77

Creating and Modifying SIP Lines in Interaction Administrator ......... 80 Line Configurations not exposed through Interaction Administrator 81 Creating A SIP Line ...................................................................... 81 SIP Configuration Page ...............................................................82 SIP Protocol Page.......................................................................86 SIP Authentication Page..............................................................87 SIP Compression Page ................................................................88 SIP Proxy Page ..........................................................................91 Registrar Page...........................................................................92

17.1 17.2

17.2.1 17.2.2 17.2.3 17.2.4 17.2.5 17.2.6

18 Defining Global Configurations SIP Stations in Interaction Administrator ............................................................................................. 93



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

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 General Page........................................................................... 100 Connection SIP Address Page .................................................... 101 Identification SIP Address Page.................................................. 102 SIP Authentication Page............................................................ 106 19.2.1 19.2.2 19.2.3 19.2.4 20

Dial Plan Basics for SIP.................................................................... 107 Dial Plan General Info ................................................................ 107 Dial Plan Verification and Testing............................................... 110

20.1 20.2 21

Gateway Configuration .................................................................... 111 Dial Plan: Configuring Gateway Selection................................... 112 Dial Plan: Configuration of Displayed Numbers ......................... 115 Example 1 .............................................................................. 115 Example 2 .............................................................................. 116 Detecting Gateway Failure and/or Congestion .............................. 117

21.1 21.2

21.2.1 21.2.2 21.3 21.3.1

Multiple Gateway Configuration ................................................. 117

21.3.2 Choosing the Proper Gateway: Configuring Gateway Selection by using an External Proxy.......................................................................... 117 21.3.3 Choosing the Proper Gateway: DialPlan 117 22 Configuring Gateway Selection by

Fax Configuration ............................................................................ 118 Availability ................................................................................. 118 Fax Detection ............................................................................. 118 Scenarios ................................................................................... 118 Inbound Scenario..................................................................... 118 Outbound Scenario .................................................................. 119

22.1 22.2 22.3

22.3.1 22.3.2 22.4 22.5

IC Server Configuration.............................................................. 119 Gateway Configurations ............................................................. 119 Cisco...................................................................................... 120

22.5.1 23

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



24 25

Tie Line and Multi-site Configuration ............................................... 122 Switchover Configuration ................................................................ 123 Switchover Component .............................................................. 123 Station Configurations................................................................ 123 Switchover in a WAN Environment ............................................. 123

25.1 25.2 25.3 26

Interaction Client Configuration ...................................................... 124 Associating the Interaction Client with a Station ....................... 124 Configuring the Interaction Client for Audio............................... 125 Special Messenger Considerations for SIP Enabled Interaction Client127 Special Server Considerations for SIP Enabled Interaction Client ..... 127

26.1 26.2

26.2.1 26.2.2 26.3 27 28 29 30

Monitoring SIP Line Activity with the Interaction Client............. 127

Phone Services ................................................................................ 128 IP Resource Management ................................................................ 132 Configuring the Message Button For Voicemail Retrieval ................. 134 Configuring Voice Mail For Non-Managed Phones (Diversion).......... 135 Logic .......................................................................................... 135 Setup.......................................................................................... 137

30.1 30.2 31 32 33

Configuring Message Waiting Indicators (MWI) .............................. 138 Configuring the Managed Phone Shortcut ........................................ 140 Sample Configurations ..................................................................... 141 Central Site Only, Primary Interaction Center Only, Cisco IP Phones 141

33.1

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 Tracing ....................................................................................... 153 No Audio Problems ..................................................................... 154 Echo ........................................................................................... 154 Audio Quality Problems .............................................................. 154 DTMF Problems .......................................................................... 155 IVR DTMF Recognition Problem .................................................. 155 No IVR, Plays, or records .......................................................... 155 DTMF from Managed Phone not being recognized by remote system 156

35.1 35.2 35.3 35.4 35.5

35.5.1 35.5.2 35.5.3 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 35.6.6 35.6.7 35.6.8 Calls made from SIP phones do not show on Line Details Page ....... 156 Phone rings when I use the MakeCall button in the Interaction Client 157 Managed station not ringing ...................................................... 157 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 Command Line Tools .................................................................. 158 Coder Bandwidth Usage ............................................................. 158 NetIQ ......................................................................................... 158 Speakeasy .................................................................................. 159 RTP Audio Monitor and Analysis Guide ....................................... 159

36.1 36.2 36.3 36.4 36.5 37

Index ............................................................................................... 160



1 Change LogThe 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. ChangeUpdated Specifying your firmware section with new table. This now appears on page 6. Typo corrections Updated firmware specifications table. Added procedure for changing firmware values. Updated DCM Network configuration settings with examples and corrected values. Updated IPLink firmware names. Added Things to watch for section with a note about not using Terminal Services or Citrix Metaframe to run Dialogic Configuration Manager. Added related documents to introduction section and added Troubleshooting section 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 Added Configuring Your System For Mu-law section, Notes About User and Station 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 Add AudioCodes setup information. Add info on SIP addresses Vendor specific portions, removed terminal services section, add MWI and message button configurations, add sip Q&A section, VAD, when changes of stations and line take affect, add pictures of topologies Hardware restrictions More Q&A charts, Outbound logic section AudioCodes update, new AudioCodes boards, new Firewall/NAT section, new Identification section in station configuration VPN, Gateway selection Added RTP Sender Report section, better incoming logic description, N+1 and redundancy, unique station and user extensions, better info on dial plans Power Usage, better description of id field Update on AudioCodes board model numbers (ver P03), SIP channel bank Q&A.

Date1/14/2002 1/16/2002 1/17/2002

1/21/2002 1/23/2002 1/24/2002

1/25/2002 2/11/2002

2/28/2002 3/1/2002 /2002 3/18/2002

3/21/2002 3/26/2002 4/14/2002 4/24/2002 5/14/2002 5/28/2002 6/3/2002 On CIC 2.2 GA CD 7/1/2002

More version numbers for Aculab, audiocodes firmware path is mandatory, Cisco SIP products Q&A, remove retired version P02 AudioCodes boards and ScBus IPM-260A120-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.



ChangeEdited content for typos. No substantive content changes. (SMS) Added EIC release directly by CIC release, new table for hardware platforms, 4.2 SIP 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) Added H.100 termination to AudioCodes Setup server parameter, better dial plan for 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 ipvs_evr_isdn_qsige1_311.pcd from 301 (15.2), fix typo in 15.5 (0x0A should be 0xA0).

Date7/22/2002 7/30/2002

8/13/2002 8/19/2002

8/30/2002

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 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 More known issues, support for Audiocodes 30 and 60 port boards, Dialogic HMP, more updates on when a proxy server is needed. Updated managed short cut info, large packet size info, reworked known issues section, firewall config, HMP issues, better diversion documentation, HMP link, better doc for switchover and station configuration Multiple NIC explanation, more work on known issues, tel scheme, more on vad, HMP fixes, identification for stations. Dialogic PTR bundle 1, Audiocodes card placement in Dialogic system, phone services, whats new section, new /mssipaudio:xxx flags 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, 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 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 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 combination: Please note that Interactive Intelligence assumes no liability with 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 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 Tell me about Ciscos skinny protocol in the Q&A section, Tie Line and Multi-site Configuration chapter, section in Audiocodes chapter about switch configuration, 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), new hot fix 1599 and 1601. New server parameters for Aculab gain control and agc, typos, dsedit parameter sections, 1633 and 1637 hot fixes Registry setting for HMP, global station dsedit parameters, more screen shots for gateway selection.

9/26/2002 On EIC 2.2 GA CD 10/18/2002 11/18/2002

12/17/2002 1/24/2003 2/28/2003

3/18/2003 4/3/2003

4/4/2003 (PL)

5/9/2003

6/3/2003 6/16/2003

6/30/2003

7/28/2003 7/31/2003



ChangeBroken RTP Disconnect Time, added T.38 chapter, add info about Audiocodes with Dialogic boards, server parameters AudioCodes Minimum Jitter Buffer Delay and 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. Echo in trouble shooting section, more info on Network and Bus gain, more fax info, disabling secondary clock master Dialogic/AudioCodes combo is certified, new features for early media and connection call warmdown time, always run wdreg_gui install, Eic_OutboundSetupParams attribute, modem configuration chapter New 8.4 section for 2.3 external audio path, Broken RTP Disconnect Time warning Combined all gateway selection into a single chapter Warmdown time of 0 is wrong, addinged Inband Transfer Enabled server parameter

Date8/11/2003

8/28/2003 10/13/2003 12/11/2003

1/20/2004 3/22/2004 10/20/2004



2 Where can I get information?2.1 Interactive Intelligence Web SiteHead 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 UpgradesInteractive 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.

Get the latest versions of software.

2.4

Whats NewWhats NewMultiple Play optimizations. Media platforms (Aculab, Intel HMP, Intel/Dialogic hardware) can be configured so that regardless of the number of calls into an

ReleaseCIC 2.2 SR-D



Communit 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 18.1Global Station Configurations not exposed through Interaction Administrator) and station configurations (see section 19.1Station 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 EIC 2.2 SR-A Communit 2.2.1 SR-C

Phone Services (section 27 Phone Services) Additional support for Actiontec and Clarisys phones (see the SIP 3rd Party Component Application Note) and section 26.2 Configuring the Interaction 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 Communit 2.2.1 GA 2.2 CIC SR-A 2.2 EIC GA Communit 2.2 GA 2.2 CIC GA

General enhancements





2.5

Known Issues with Interaction Center Products

2.5.1

Hot FixesHot Fixeshttp://www.inin.com/support/cic/22/updates/indexsrd.asp?q=880 http://www.inin.com/support/cic/22/updates/indexsrd.asp?q=880 (2.2.2 is based on 2.2 SR-D) http://www.inin.com/support/eic/22/updates/index2.asp?q=830 You should update to the most current service release above. You should update to the most current service release above. You should update to the most current service release above. You should update to the most current service release above. You should update to the most current service release above. You should update to the most current service release above. You should update to the most current service release above. You should update to the most current service release above.

ReleaseCIC 2.2 SR-D Communit 2.2.2 GA EIC 2.2 SR-A CIC 2.2 SR-C Communit 2.2.1 SR-C 2.2 CIC SR-B Communit 2.2.1 GA 2.2 CIC SR-A 2.2 EIC GA Communit 2.2 GA 2.2 CIC GA

2.5.2

Known Issues

All issues in the most recent releases are issues in previous releases, unless noted.

Issue

Workaround

Affected Releases If no *, then affects this release plus prior releases

Release Fixed In

Hot Fixes

CIC 2.2 SR-DDouble digit problem on last digit on an internal call. The last dialed digit might be taken as the first digits as well when in the IVR. CIC 2.2 SR-C EIC 2.2 GA

2.6Issue

Known Issues with Other ProductsWorkaround Affected Releases Release Fixed In Hot Fixes

General



Some SIP devices do not understand delayed media. Delayed media is used for outbound calls when over one codec is configured in the line.

You can change from delayed media (new feature and new default) to normal media (previous default) by only selecting one codec in IA (see section 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.

CIC 2.2 SR-C EIC 2.2 SR-A

No fix necessary. Most devices handle delayed media.

Microsoft MessengerSome versions Microsoft Messenger will try to use an odd port number for audio. This is not valid with HMP or AudioCodes.

Cisco VPN SoftwareMicrosoft Messenger does not work with Cisco VPN software. The Cisco VPN does not expose its interface, thus Messenger passes internal IP addresses in its SIP messages (SDP and 200 OK). Use Microsofts PPTP VPN software rather than Ciscos VPN software. Ciscos 4.0 VPN.

ActionTecActiontec phones only: A buzz is heard by remote caller when an actiontec phone goes offhook to answer a call. None. CIC 2.2 SR-C* EIC 2.2 SR-A* * Actiontec support is new in these releases.



3 Glossary of TermsManaged 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 IntroductionWith 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 its 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 NotesDialogic 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.



4.24.2.1

StandardsOther CompaniesInteractive 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 RFCThe 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 RFC 2543bis04 RFC 2327 Session Description Protocol RFC 2617 Basic and Digest Access Authentication2.2 EIC SR-A, 2.2 CIC SR-C with hot fix

RFC Drafts draft-ietf-sip-refer-022.2 EIC GA, 2.2 CIC SR-B, Communit 2.2.1

Description Session Initiation Protocol. Description of the session within the SIP messages Only Digest Access Authentication is supported. Basic Access has been deprecated by RFC3261 (SIP) and is not supported. Description REFER

draft-ietf-sip-refer-072.2 EIC SR-A, 2.2 CIC SR-C

draft-biggs-sip-replaces-01 draft-ietf-sip-cc-transfer-05 draft-levy-sip-diversion-03 draft-mahy-sip-message-waiting-02 draft-ietf-sip-service-examples-03 draft-ietf-sip-events-05 Coming soon. RFC 3261

Replaces Consult Transfer (uses REFER/Replaces) Blind Transfer (uses REFER) Voicemail for unmanaged phones (uses Diversion/CC-Diversion) MWI (uses SUBSCRIBE/NOTIFY) Hold SUBSCRIBE/NOTIFY Description 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 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.

draft-ietf-avt-rtp-cn-06

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 PortsIP Address Used IP Address of the systems NIC (Network Interface Card) For the hardware platforms, the IP Address of the IP telephony board.. For software platforms (HMP), the IP Address of the systems NIC (Network Interface Card) For the hardware platforms, the IP Address of the IP telephony board. For software platforms (HMP), the IP Address of the systems NIC (Network Interface Card) IP Address of the Interaction Center system IP Address of the Interaction Center system Port Number Used 5060. This is configurable. 4000. This is port number of the first RTP session. This is configurable. The second RTP session will start at an even port number higher than 4000. This will always be one higher than the port number used for its RTP session.

Protocol SIP RTP

RTCP

Interaction Center Notifier Interaction Center Web Services

2633, registered with IANA (http://www.iana.org/assignments/portnumbers ). 3508, registered with IANA (http://www.iana.org/assignments/portnumbers ).

4.2.6

Security AlertThere 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&ADoes EIC, CIC, Communit, Vocalit , Mobilit , 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? Ive 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 Ciscos skinny protocol (called SCCP), H.323, and SIP. How do their interrelate? Here is a little Q&A:

Why aren't Ciscos 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 Center with the Cisco TAPI Platform Interaction Center with SIP

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



How does this compare to running the TAPI version of the Interaction Center?TAPI vs. SIP comparison Can the Interaction Center system be mixed with traditional connections via telephony boards, such as analog phones and ISDN trunks? What features are lost? TAPI Interaction Center No. Interaction Center with SIP capabilities Yes. We simply added SIP to our telephony board version.

A few due to the restrictions of the TAPI interface that Cisco provides. See the TAPI app note for details. Yes. Yes. We have a single vendor dependency on Cisco. This typically leads to higher cost equipment.

None.

Is a Cisco CallManager needed? Hardware does all the gateways, routers, and phones have to be from Cisco?

No. No. Multi vendor solutions are used. The first gateways and phones we 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. All SIP.

Software are proprietary or standard protocols used to communicate with phones and gateways.

Proprietary. Mixture of standard and proprietary protocols.

So, in SIP terms, what are you?SIP component Application Application Server Media Server User Agent Client User Agent Server Proxy Registrar SIP Gateway Does the Interaction Center have the features of this SIP component? Yes. Yes. Yes. Yes. Yes. Yes or we can work with any SIP compliant SIP Proxy. Yes or we can work with any SIP compliant SIP Registrar. 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 Must be a hard phone. Takes up a dedicated station resource (either a station port on a station board or a T1/E1 channel when using a channel bank). Must be locally connected to the server or to a channel bank. Flash is used to hold or bring up a voice menu to do features like conference and transfer.

SIP Phone Can be either a hard or soft phone. Uses a resource only when in use.

Can sit anywhere on the LAN or WAN.

Some IP phones have buttons to do hold, transfer, conference,

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 agents Station in Interaction Administrator. See section 18 Defining Global Configurations SIP Stations about configuring Persistent connections.



4.44.4.1

Implementation Overview DiagramsPicture: SIP Hardware Approach OverviewThe 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 VoIP Call Control (SIP) SIP Stack Interaction Center Software Net wo rk Ca rd Vo IP Audio (RTP) IP Cards Telepho ny Bus Resources (fax, confe rencing, audio) Analo g Station Cards ISDN, T1, E1, Ana log PSTN WAN IP LAN

SIP Soft and Hard Phones

Gateway/Routers PSTN Analog Phones

Tele phony Code

SIP Soft and Hard Phones

Trunk Cards

5 When is a SIP Proxy NeededThis 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 RoutingSome SIP Phones

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



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

SIP Proxies can be used to route SIP Messages. The Phones, Gateways, and Interaction Centers can use the Proxy for all the routing decisions.

LANSIP MSIP Phones and Gateways

e s s ag

e P at

Interaction Centersh

SIP Proxy Server

Some IP Phones can route SIP Messages. If the specific phone can not route SIP messages, then a SIP Proxy must be used.

LAN Interaction CentersSIP Message PathSIP Phone

Some Gateways can route SIP Messages. If the specific gateway can not route SIP messages, then a SIP Proxy must be used.

PSTN / WANSIP Gateway

LANSIP M essa

ge P a

Interaction Centersth



5.2

Phone Specific RoutingSome 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 dont 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 phones 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 phones 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 dont 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 phones 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 OverviewThe 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 Detail7.1 PSTN Connectivity OptionsSee [1]

One or all of the options below can be mixed on same system!!! 1. No gateway (tradition connections, such as ISDN, from carrier). 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

1

PSTN

ISDN, T1, E1, Analog

2

PSTN / WAN

ISDN, T1, E1, Analog

3

PSTN / WAN

SIP

LAN

4

PSTN / WAN

SIP

LAN

1IC Servers with no gateways, using ISDN connections to the PSTN Gateway Features No gateway. PSTN connectivity is done via the telephony boards.

2IC Servers with ISDN connections to gateways

3IC servers with SIP connections to gateways

4IC Servers with no gateway, using SIP connections to PSTN/WAN

Connect to the PSTN and WAN via tradition connections (ISDN, Frame Relay) and then connect to the IC server via traditional connections (ISDN,). Tradition ISDN (or T1, E1, Analog) telephony boards are used to connect to the gateway.

Connect to the PSTN and WAN via traditional connections (ISDN, Frame Relay) and then convert all traffic to SIP.

No gateways necessary. PSTN and WAN connectivity is done via SIP. This is not available yet, but is coming soon by large carriers.

Are Telephony boards needed?

Tradition ISDN (or T1, E1, Analog) telephony boards are used to connect to the PSTN.

Optional. With the hardware platform (telephony boards), IP boards are used to do the do the RTP and transcoding. With the software platform (Intel HMP),

Optional. With the hardware platform (telephony boards), IP boards are used to do the do the RTP and transcoding. With the software platform (Intel HMP),



1IC Servers with no gateways, using ISDN connections to the PSTN For switchover (primary and backup IC servers), is a data probe needed to route the digital lines? Yes. The traditional connections (such as ISDN) go through the data probe, which routes the connections to the appropriate server. The calls are distributed, by the PSTN, across the IC servers, by sending the call to different ISDN trunks.

2IC Servers with ISDN connections to gateways

3IC servers with SIP connections to gateways

4IC Servers with no gateway, using SIP connections to PSTN/WAN

Yes. The traditional connections (such as ISDN) go through the data probe, which routes the connections to the appropriate server. The calls are distributed, by the gateways, across the IC servers, by sending the call to different ISDN trunks.

No. All connections to the IC server are done via SIP. With SIP, the switchover routing is done over the LAN.

No. All connections to the IC server are done via SIP. With SIP, the switchover routing is done over the LAN.

N+1 Configuration (multiple IC servers)

The calls are distributed, by the gateways, across the IC servers, simply by sending the SIP messages to different IP addresses.

The calls are distributed, by the PSTN, across the IC servers, simply by sending the SIP messages 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. See [3] below. 3. Media Gateways.

Analog Phones or PBX Digital Phones

Phone Media Gateway IP LAN

SIP Co mpliant Soft Phones with or without Interaction Client SIP Co mpliant Hard Phones with or without Interaction Client Interaction Client used for Audio

3 2Interaction Center

1Analog Phones

2IP WAN

SIP Co mpliant Soft Phones with or without Interaction Client SIP Co mpliant Hard Phones with or without Interaction Client

3Analog Phones or PBX Digital Phones

Phone Media Gateway

Interaction Client used for Audio

1IP Phones Is SIP used to communicate to the phones Yes.

2SIP Phone Media Gateways Yes. The IC server communicates with the Phone Media Gateway with SIP. The Phone Media Gateway then communicates with the phone the same way a traditional channel bank does. No. IP resources are only used when there is a voice connection. No. The Phone Media Gateway is simply an IP device anywhere on the network (LAN or WAN).

3Analog Phones No. Tradition T1/E1 boards for channel banks, or analog station boards are used to connect to analog stations. Yes. The phone uses a physical resource even when it is idle. Yes. The phone has a physical connection to the IC server.

Are resources used when phone is idle? Does the phone have to be directly connected to IC server? Phone Types supported

No. IP resources are only used when there is a voice connection. No. The SIP hard or SIP soft phones are simply IP devices anywhere on the network (LAN or WAN). Many vendors make SIP hard and SIP soft phones.

Standard analog phones (2500 sets) and PBX digital phones can be connected to a wide variety of Phone Media Gateways.

Standard analog phones (2500 sets).



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 phones 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, lets understand Ciscos 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

Ciscos NON-SIP SRST (Survivable Remote Site Telephony)

Proprietary SRST Overview

Central Site with Cisco CallManagers

Remote Site WAN 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 routers 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

Ciscos SIP SRST (Survivable Remote Site Telephony)

Ciscos SIP SRST Overview

Central Site with Interactive Intelligences Interaction Centers

Remote Site WAN LAN

PSTN

SIP capable SRST Cisco Router

SIP Proxy (optional)


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 routers 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 Intelligences Remote Survivability using SIPEven Ciscos

Again, using SIP provides the flexibility of equipment and vendors. routers support SIP.Standard SIP Approach for Remote Survivability Central Site with Interactive Intelligences Interaction Centers Remote Site WAN LAN

PSTN

SIP Router



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 routers 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 Interactive Intelligences Interaction Centers

Remote Site WAN LAN

PSTN

SIP Router



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.



7.47.4.1

Understanding the Audio PathRemote 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 Sizing8.1

IP ResourcesAn active SIP connection from a gateway (typically an external call). An active SIP connection from the Interaction Center to a managed phone. 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.

Each IP session will use an IP resource. An IP session is either:

Examples

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 agentwait-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 A to IC IC to A AudioPath Internal or External In

SIP Application Note

Documents