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Cisco Systems Response To VoiceCon2007 RFP

©2005 TEQConsult Group 1 of 215

Cisco Systems Unified

Communications Proposal

Based On:

VoiceCon Request for Proposal for an IP Telephony System

Prepared by Allan Sulkin

President, TEQConsult Group teqconsult.com

VoiceCon Spring 2007

IPTS RFP Workshop March 5, 2007



Preface

The following RFP document was exclusively designed and developed by TEQConsult for the VoiceCon® Spring 2007 Conference. The RFP is intended to solicit product information and pricing data about IP Telephony systems during the Fall 2006 time period. The RFP was written for a large multi-facility enterprise configuration with IP voice terminals as the primary station user interface to the system. TEQConsult Group recognizes that every business and institution has unique communications needs and resources, but the much of the material included herein can be used by VoiceCon workshop attendees regardless of their unique system size and application requirements. VoiceCon workshop attendees may use this RFP as a template for customizing their own RFP with the proviso that proper accreditation to TEQConsult Group will be included in the document. TEQConsult Group would like to thank Fred Knight, VoiceCon GM and the publisher of Business Communications Review, for his review and editing of this document, and to Unimax Systems Corporation for its contributions to the systems management section of the RFP.



Cisco Legal Disclaimer

Cisco Systems, Inc. (“Cisco”) is extremely pleased to present this proposal for your evaluation and consideration. Please note that the information contained in this proposal is proprietary and confidential to Cisco, and is furnished in confidence to you with the understanding that it will not, without the express written permission of Cisco, be used or disclosed for other than proposal evaluation purposes. This proposal is not, and should not be construed as, an offer to contract with Cisco. If you ultimately decide to purchase any or all of the products and/or services described in this proposal directly with Cisco, then all terms and conditions (inclusive of all business terms and conditions) will only be pursuant to a final and definitive written agreement, in the form of either: (i) Cisco’s standard U.S. Terms and Conditions of Sale (a copy of which is available at: www.cisco.com/legal), (ii) an existing written agreement between us, or (iii) a mutually negotiated final written agreement. For purposes of clarity, for a direct relationship with Cisco, the final agreement would replace any other suggested terms and conditions, and Cisco hereby takes exceptions to any such purported terms and conditions. Notwithstanding anything to the contrary, Cisco makes no representations, warranties or covenants in this proposal (including without limitation as to any products, services, service levels, third-party products or services or interoperability) separate from, in contravention of, or in addition to those contained in the final agreement, and any purported representation, warranty or covenant in this proposal shall be of no force or effect. If you desire a direct relationship with Cisco, we would welcome the opportunity to discuss mutually acceptable terms and conditions. Alternatively, you may choose to purchase the Cisco products and services through a Cisco authorized reseller, and the terms and conditions, and all pricing, would be governed by your contract with such reseller. Cisco cannot, in any fashion, dictate or control resale pricing. For further information about Cisco’s authorized resellers, please see:

www.cisco.com/en/US/partners/index.html

Any information contained in this proposal relating to pricing or to future technology under development may be subject to change, including as a result of the negotiations which might occur in contemplation of the final agreement. If any pricing is provided by Cisco in this proposal, it is provided solely for your convenience and budgetary purposes only, and does not constitute a bid or an offer from Cisco. Any other pricing will be provided directly by an authorized reseller, and any discussions relating thereto should be held directly with such reseller and not Cisco. Any descriptions, documentation or references to third party products not on Cisco’s price list are provided for informational purposes only and shall not be considered a part of Cisco’s proposal. Thank you for considering Cisco for this exciting opportunity. We look forward to further assisting you with your technology requirements.



PART 1: System Performance Requirements Submit Part 1 responses in MS Office WORD file format except when otherwise noted. 1.0.0 System Overview The VoiceCon Company plans to install a new IP Telephony System (IPTS) network to support its newly constructed Headquarters (HQ) facility, a Regional Office (RO), and three Satellite Branches (SBs) with Survivable Remote Gateway (SRG) capabilities. Dedicated local IPTS call telephony servers must be installed at the HQ and RO facilities. All proposed call telephony servers must independently support all generic software features for the proposed IPTS model(s) as required in Section 5 of this RFP. The three SBs will be configured as survivable remotes behind the HQ IPTS call server with local trunk circuit services (Note: Survivability requirements for the SB facilities are identified later in this section). The proposed IPTS network solution may include a single fully distributed IPTS or no more than two IPTSs (each housed at HQ and RO facilities). If a single IPTS is proposed the distributed call servers must function and operate independently of each other, and support all generic software features as required in Section 5 of this RFP. The HQ IPTS call server will initially support 1,360 station users at the HQ and three SB facilities. The RO IPTS call server will initially support 250 station users. See Figure 1 for an overview of the VoiceCon IPTS network. See Figures 2 – 6 for port capacity requirements at each of the five VoiceCon facilities. VoiceCon anticipates 50% station user growth at the HQ and RO facilities, only, and the proposed IPTS network solution must accommodate this growth without replacement of any installed hardware/software. There is no anticipated growth at the SB facilities. A centralized messaging system will be housed at the HQ facility and must be capable of supporting station users located at all VoiceCon facilities (HQ, RO, and SBs).



HQ IPTS1200 stations

RO IPTS250 Stations

SB1 SRG100 Stations

SB3 SRG10 stations

SB2 SRG50 Stations

WAN

Figure 1

Voicecon IPTS Network

HQ: HeadquartersRO: Regional OfficeSB: Satellite BranchIPTS: IP Telephony SystemSRG: Survivable Remote Gateway



HQ IPTS1200 stations

6 Local T1 circuits7 Long Distance T1 circuits

5 PFTS circuits25 Emergency Analog GS/LS Circuits

Figure 2

HQ Port Requirements

RO IPTS250 stations

2 Local T1 circuits2 Long Distance T1 circuits

2 PFTS circuits10 Emergency Analog GS/LS Circuits

Figure 3

RO Port Requirements



SB1 SRG100 stations

1 Local T1 circuit2 PFTS circuits

5 Emergency Analog Circuits

Figure 4

SB1 Port Requirements

SB2 SRG50 stations

10 Analog LS/GS circuits2 PFTS circuits

Figure 5SB2 Port Requirements



SB3 SRG10 stations

5 Analog LS/GS circuits1 PFTS circuit

Figure 6SB2 Port Requirements

VoiceCon has plans to install at all of facilities LAN/WAN cabling and a transport infrastructure that will fully satisfy the stringent requirements of IP Telephony communications for all intra-premises and inter-premises call control and voice communications transmissions. Each location will be equipped, at minimum, with a 1-Gbps Ethernet backbone. The local wiring closets will house 10/100/1000 Mbps Ethernet switches equipped with Power over Ethernet (PoE). Multi-service routers will be installed at all locations to support a MPLS WAN installation. All Ethernet switches and IP WAN routers will be equipped and programmed to satisfy QoS and security standards necessary to support voice communications acceptable to VoiceCon. Pertinent bandwidth, latency, packet loss, and echo issues will be addressed in the design and implementation. Each station user’s work area will be supported by four (4) four-pair, Category 5E cable wiring with one (1) RJ-11 wall connector and three (3) RJ-45 wall connectors to the local wiring closet. The RJ-11 and RJ-45 connectors will be either wall mounted or mounted in the modular furniture throughout the office environment. VoiceCon plans to run its IP Telephony system over this cable infrastructure. NOTE: The proposed IP Telephony system must be able to support a limited number of non-IP stations, e.g., analog telephones, requiring a RJ-11 connector. The proposed system can use either circuit switched port carriers or media gateways to support analog communications terminal equipment.



Vendor Response Requirement Based on the RFP requirements in this document prepare a simple network diagram that illustrates the proposed IPTS network design. Include in the diagram the brand name/model of the IPTSs, all circuit switched port carrier/media gateway equipment, the brand/name of the HQ-located systems management and messaging system. The diagram must be prepared and submitted in a separate file using MS PowerPoint format (identify the file as part of your electronic proposal submission). In addition copy/paste diagram in the submitted MS WORD file proposal here.

Cisco Response: The proposed Cisco IP communications solution is shown in the diagram below. The details of the configuration are fully described in Sections 1.2.0 and 1.3.1:

1.0.1 LAN/WAN Requirements VoiceCon has not yet decided on the make/manufacturer of its new LAN/WAN communications equipment.



Vendor Response Requirement Indicate if the proposed IPTS solution for the HQ and the remote facilities requires manufacturer-specific LAN/WAN communications equipment to support any or all of the following voice communications operations or functions: call processing, switching, routing, PoE, media gateway, QoS and security. If responding in the affirmative, only, identify the make and model of the necessary switch/router equipment and the reason for its requirement.

Cisco Response: Read and Understood. Cisco Unified IP communications can be deployed on any standards-based, QoS-enabled switched Ethernet infrastructure that is properly configured. For example it is assumed that the data infrastructure supports 802.1p/q, multiple queues, IEEE standard PoE, etc. Therefore, there is no requirement that the underlying LAN/WAN be based on Cisco Products. This RFP response assumes that the underlying data infrastructure at VoiceCon could be supplied by any vendor as long as it meets the requirements stated above. That said it is Cisco’s position that a higher degree of functionality is achievable for IP communications when deployed on Cisco data infrastructure components. Cisco call processing and other applications, and Cisco Unified IP Phones can take advantage of tight functional integration with Cisco data equipment to enhance QoS, security, and manageability as well as to provide value added capabilities such as automated E911 support. In addition, VoiceCon could leverage the “self-defending” capabilities of the underlying Cisco data infrastructure for IP communications security.

1.1.0 Basic IPTS Requirements The proposed IPTS equipment should be in current production and operating as part of a commercial system for at least five (5) customers in the USA. Vendor Response Requirement State if the proposed IPTS equipment satisfies this commercial availability requirement. If the IPTS model has not yet been shipped and installed in a commercial installation, state expected availability date. Also provide an estimate of the number of IPTS solutions (same model as proposed) currently installed and operating in the USA.

Cisco Response: Read and Understood. Cisco complies. As of the date of submission of this RFP response Cisco has over 48,000 customers worldwide, has shipped more than 10 million Unified IP Phones, and installed over 6.6 million Unity seats and 1 million Contact Center seats. Cisco has more than 150 customers who have deployed systems with more than 5,000 IP phones.

NOTE: All proposed system hardware and software must be formally announced as of VoiceCon Spring 2007 to be accepted by VoiceCon in response to this RFP. This is a mandatory requirement to submit a



RFP response.

Cisco Response: Read and Understood. Cisco complies. All hardware and software versions referenced in this document will have been announced as of VoiceCon Spring 2007.

1.1.1 Single System Image The proposed IPTS network should provide a Single System Image across VoiceCon HQ, RO and SB facilities. The Single System Image should include, but not be limited to, the following: 1) 5-digit dialing between all station users; 2) High degree of transparent operation across all VoiceCon facilities for station, attendant, and system features (see RFP Section 5: Call Processing Features); 3) HQ-located centralized systems management solution using a single unified database for all station user profiles, equipped system design, and system-level operations; 4) Network-wide attendant operator services across all VoiceCon facilities, including the ability to support a centrally located attendant pool; 5) Shared messaging system resources; 6) Automatic alternative routing across the network for all voice calls (station-to-station and PSTN trunk connections). Vendor Response Requirement: Provide specific answers to each of the following questions: 1. Is the proposed IPTS network solution a single system solution or multiple systems intelligently networked? 2. Does the proposed IPTS network solution fully satisfy all six (6) of the stated Single System Image requirements? If not, explain which of the requirements are not satisfied?

Cisco Response: The proposed Cisco Unified CallManager system is a single system solution. It fully satisfies all six of the stated single system image requirements. Cisco has proposed a centralized deployment model with the redundant Unified CallManager cluster divided across the primary headquarters and the regional office locations. The Cisco Unified CallManager system uses clustering over the WAN to provide full redundancy across locations. In addition, the servers are fully redundant at each location. Servers can back each other up across the WAN link, and if that link is lost each site can operate independently (with full redundancy still at each site as well). Even though the system utilizes multiple servers within the cluster, it provides a “single system image” across all locations, with full feature transparency to the regional offices and the branches. The proposed system fully satisfies all six of the stated single system image requirements. All supported features and applications such as, Unity Unified Messaging are equally available to all users on the system. Automatic Alternate Routing between WAN and PSTN



connections is a standard feature. In addition, management is centralized at the Unified CallManager cluster. In the future, VoiceCon could install Cisco Unified Contact Center Enterprise at the primary headquarters location and provide a fully distributed contact center solution across all locations on the same infrastructure.

1.1.2 Enhanced 911 (E911) Services Support It is mandatory that the proposed and installed communications system support E911 services provided by a public safety answering point (PSAP) as defined by FCC regulations. All VoiceCon IPTS network locations addressed by this RFP are served by the same PSAP. All VoiceCon IPTS station user E911 calls must be directed to their local PSAP for call handling and response regardless of location, i.e., facilities remote from the primary call telephony server. If more than one E911 solution is available for the proposed IPTS network configuration clearly specify the solution that is included in the price proposal. Vendor Response Requirement: Confirm that the proposed communications system solution supports E911 service for all user stations (IP and analog) at each of the VoiceCon facilities. In the response briefly explain how E911 service requirements are supported, specifically addressing each of the following questions:

1) A description of any optional hardware/software equipment included in the pricing proposal, and if a peripheral server is required who is responsible for its purchase?

2) How are station user moves/adds/changes reported to the E911 provider?

3) What degree of specificity station user location is identified to the E911 PSAP? Desktop work area, local switch room, work floor, other?

Cisco Response: Cisco Unified CallManager fully supports E911 without the need for any additional or optional hardware or software. E911 basically involves routing calls to the appropriate PSAP with specific ANI information (ELIN) that is used to determine the caller's location (ERL) and the ability to call that number back and have it ring in the vicinity of the original 911 caller or to a security desk. Cisco Unified CallManager handles this scenario through the use of calling search spaces, partitions, route patterns and translation patterns. Adds/moves/changes must be manually tracked to determine if a caller’s location has changed to a different ERL which will require a change to the phone's calling search space and partition configuration.

Therefore, in a standard configuration Cisco Unified CallManager supports E911 calls in the same manner as a legacy PBX system. That is, as long as manual updates are made to the PSALI database to reflect phone moves and changes, E911 calls will be properly routed to the PSAP and provide accurate



location information. However, when coupled with the optional Cisco Emergency Responder application Cisco can offer an E911 solution with unprecedented functionality. On a Cisco data infrastructure of Catalyst switches you can leverage the underlying intelligent services such as, CDP to automatically track phone movement. On a non-Cisco data infrastructure phone movement can be tracked via IP subnet. This allows you to implement Cisco Emergency Responder in a switched Ethernet environment where the data switches are provided by different vendors. Cisco Emergency Responder is optional and runs on an adjunct Cisco MCS server. Cisco Emergency Responder has been included in this proposal and is included in the total system price under the “Optional Software” category. The following table is a breakdown of the list price of Cisco Emergency Responder to support VoiceCon’s configuration: This application is very cost effective. At list price for this proposal, it costs $34,294 (See pricing summary under “Optional Software”). This adds about $21 per line to support VoiceCon’s 1,610 user configuration, and provides a fully automated E9-1-1 solution that provides near real-time updates of location information. The following section includes a more detailed description of Cisco Emergency Responder: Cisco Emergency Responder enables emergency agencies to identify the location of 911 callers and eliminates the need for any administration when phones or people move from one location to another. Enhancing the existing E9-1-1 functionality of Cisco Unified CallManager, Cisco Emergency Responder’s real-time location-tracking database and improved routing capabilities direct emergency calls to the appropriate Public Safety Answering Point (PSAP) based on the caller’s location. When coupled with Cisco Unified CallManager, Cisco Emergency Responder surpasses traditional private branch exchange (PBX) capabilities by introducing zero-cost user or phone moves and changes and dynamic tracking of user and phone locations for E9-1-1 safety and security purposes.

Feature Overview

Automatically Locates Phones and Users Cisco Emergency Responder proactively queries Cisco Unified CallManager for new phone and user login registration events. In response to these events, Cisco Emergency Responder automatically searches known Cisco Catalyst® switches in the network and finds the location of the phone and the user, based on the switch port to which the phone is attached. This information is then updated in a Cisco Emergency Responder location database, and is used to identify a caller’s location when an E9-1-1 call is placed. With this solution, users can move within a campus or between sites, wherever and whenever they want, without any administrative intervention from the IT organization. This eliminates the administrative costs associated with relocating phones or users, while maintaining accurate and updated location information for E9-1-1 state and safety mandates.



Dynamically Routes Emergency Calls and Provides Real-time Location Information Cisco Emergency Responder makes informed inbound and outbound call-routing decisions based on the location of emergency callers, and provides crucial location information to emergency operators in Public Safety Answering Points (PSAPs). Outbound emergency calls are directed to a gateway associated with the PSAP that is nearest to the caller, and in the event of an unintentional call disconnect or need for additional information, inbound calls from a PSAP are returned to the original caller. For example, Dallas telephone users who visit Chicago and dial 911 are connected to a PSAP in Chicago, even though a Cisco Unified CallManager cluster in Dallas processes their calls. The Chicago PSAP receives accurate and updated location information about the 911 callers visiting Chicago from Dallas, and can return their calls. When the Dallas users return home, their subsequent 911 calls are directed to a PSAP in Dallas, with no administrative intervention from the IT organization. Again, the Dallas PSAP receives accurate and updated location information, and can return their calls.

Cisco Emergency Responder achieves this breakthrough in E9-1-1 administration by transcending the traditional method of a user’s phone number to a physical location. Rather than sending the caller’s phone number in the calling party number field of outbound emergency calls, Cisco Emergency Responder sends a different Direct Inward Dial (DID) number that represents the current physical location of the caller. This substitute DID number, called an Emergency Location ID Number (ELIN), acts as a key to the location database which local exchange carriers and PSAPs use to route calls and identify caller location. Data from physical floor plans and site cabling plans are posted a single time to a Private Switch Automatic Location Identification (PS-ALI) database, and no additional updates are required for any user moves, adds, or changes.

This elegant solution both meets and exceeds traditional E9-1-1 requirements. User and phone location changes are automatically updated in real time, whereas traditional E9-1-1 requirements stipulate an update within 24 to 48 hours.

Provides Real-time Alerts During an emergency situation, a reduction of just a few seconds in response time can have an enormous impact on life, health, and property. Cisco Emergency Responder helps minimize response time by providing real-time alerts to onsite personnel through E-Mail, pager, telephone call, and Web page notifications. Onsite response personnel then have knowledge of a caller’s location, the owner of the originating phone, and the phone number (ELIN) as received by the PSAP. This information facilitates an immediate response before public fire, police, or medical services reach the scene, and improves the effectiveness of public services when they arrive.

Provides Auditing and Reporting Cisco Emergency Responder tracks and logs administrative changes that affect the emergency location database. This information “audit trail”



facilitates a responsible change management process, and is a valuable tool to maintain service availability. In addition, the configuration audit trail is a source of information for investigative or legal proceedings in cases of intentional misuse. Cisco Emergency Responder also maintains a commented history log of all emergency calls, which facilitates capacity planning for emergency voice trunks, monitoring of emergency call abuse, and documentation of emergency incidents.

Key Benefits Cisco Emergency Responder enhances the E9-1-1 functionality inherent in Cisco Unified CallManager, and creates an unrivaled “Dynamic E9-1-1” solution for organizations with multi-line telephone systems:

• Meets and exceeds traditional E9-1-1 requirements • Automates all user and phone moves, adds, and changes • Enables users and phones to move an unlimited number of times per

day • Avoids the expense and burden of daily PS-ALI record uploads • Avoids daily documentation updates that potentially introduce errors • Enables quicker and more effective emergency response from onsite

personnel and public agencies • Provides configuration auditing to facilitate responsible change

management and investigative or legal processes • Provides call history logs for capacity planning, management of

emergency call abuse, and incident documentation • Is compatible with any emergency number (for example, 999 in UK,

E1-1-2 across Europe) As a result, customers experience cost savings from reduced telecom administration of user moves, adds, and changes. And by freeing telecom resources from tedious E9-1-1 location database updates, customers can be more productive and focus more of their attention on core strategic initiatives. Because ongoing maintenance of Cisco Emergency Responder is minimal, E9-1-1 compliance is converted from an administrative and financial burden to an opportunity for organizational gains through risk management, loss prevention, site security, employee safety, and community stewardship. System Capacity A single Cisco Emergency Responder server supports a full Cisco Unified CallManager cluster of 30,000 phones and 2,000 LAN switches (120,000 ports) with attached phones in the same PS-ALI database reporting area. Cisco recommends a second Emergency Responder server to form a fully redundant Cisco Emergency Responder Group with the same capacity and increased availability compared with a single Cisco Emergency Responder server. A single Cisco Emergency Responder Group can support one or more Cisco Unified CallManager clusters, provided that the total number of phones and LAN switches with attached phones is within the system capacity. Larger campuses and distributed systems are supported via a network of Cisco Emergency Responder groups called a Cisco Emergency Responder Cluster.



1.1.2.1 E911 and Station Moves It is required that station user moves behind the proposed IPTS solution be tracked dynamically in real time for E911 services support. Vendor Response Requirement: Indicate if the proposed E911 solution satisfies this desirable capability and indicate how often the database updated. If an alternative E911 solution is available that satisfies this capability, but is not included as part of the overall IPTS solution and pricing proposal, briefly describe this option and the incremental costs to purchase and install beyond the proposed solution.

Cisco Response: As described above, station moves are automatically and dynamically tracked in real time with the optional application, Cisco Emergency Responder (CER). Without CER tracking is manual and the update schedule would depend on the system administrator.

1.2.0 Proposed Communications System Design The proposed communications system may only be based on either of the two following architecture technology designs:

• Single system design based on true peer-to-peer distributed call processing topology, i.e., identical or similar call telephony servers located at all VoiceCon facilities (HQ, RO, SBs)

• Intelligently networked multiple system design based on identical or similar call telephony servers located at VoiceCon HQ and RO facilities, and survivable remote gateways at VoiceCon SB facilities configurable behind the HQ call telephony server.

Only a supplier’s most current generation hardware/software solution will be acceptable. No refurbished equipment is acceptable.

Cisco Response: Read and Understood. Cisco has proposed a centralized call processing configuration with a Unified CallManager cluster divided across the primary headquarters and regional office locations, providing call processing and applications support for all locations including the Remote Offices. The proposal is based on the currently available generation of products and all equipment quoted is new. This configuration is described in detail immediately below. All required hardware and software has been included for a complete turnkey system.

NOTE: There is no preference for either the single or multiple system design if all 1.1.1 Single System Image requirements are satisfied. Vendor Response Requirement:



Briefly describe your proposed solution, referring to the diagram from RFP Clause 1.0.0 when applicable. Limit your response in this section to the following high level information as details are requested in following sections:

1. Product and model name(s) for the IPTS(s) and messaging system. 2. Identify proposed solution as a single system or multiple system design. 3. For each network location specify the product/model used to support station/trunk call processing and switching operations under normal operating conditions. 4. Identity the software release for each product/model proposed 5. Provide the product/model introduction dates. Cisco Response: Read and Understood. Cisco's response to this proposal is structured to clearly delineate the hardware and software costs associated with installing a Cisco IP communications solution onto an existing QoS-enabled LAN/WAN data infrastructure with inline power. Since the LAN and WAN are already in place, the solution involves adding voice components such as call processing and application servers, analog and digital gateways, Unified IP Phones, and voice mail. The proposed configuration for VoiceCon is shown in the diagram in section 1.0.1 above. It is a single system with centralized call processing supporting the Headquarters, regional office, and all remote sites.

The following Cisco equipment and applications were added to provide the complete solution: Headquarters Location: The primary headquarters location contains several Cisco AVVID components:

• Call Processing and Application Servers Unified CallManager Release 5.1 Two (2) Cisco Media Convergence Servers (MCS) 7835-H2

servers. These servers are redundant, load sharing call processing servers.

One (1) MCS 7845-H2 server for Cisco Unity Unified Messaging and Automated Attendant (72 ports).

One (1) MCS 7825-H2 server for Cisco Emergency Responder • Cisco Unified IP Phones

Cisco Unified IP Phone 7906G – 100 Cisco Unified IP Phone 7931G – 140 Cisco Unified IP Phone 7961G - 650 Cisco Unified IP Phone 7970G - 50 Cisco 7936 IP Conference Station – 5 Cisco IP Communicator (softphones) – 100

• PSTN Gateways Four (4) Cisco Integrated Service Routers, Model 2821.



o Seven (7) VWIC-2MFT1 – 2-port T1 modules, providing 14 T1 ports

o Seven (7) VIC2-4FXO modules for a total of 28 FXO ports. • Analog FXS Gateways

One Voice Gateway (VG) 248 to provide 48 FXS ports for analog phones, modems, fax machines, etc.

One ATA-186 2-Port FXS Adaptor to provide 2 FXS ports • Power Failure Transfer Unit

One (1) Gordon Kapes Power Failure Bypass Unit (8 ports)

Regional Office Location: The primary headquarters location contains several Cisco AVVID components:

• Call Processing and Application Servers Unified CallManager Release 5.1 Two (2) Cisco Media Convergence Servers (MCS) 7835-H2

servers. These servers are redundant, load sharing call processing servers.

• Cisco Unified IP Phones Cisco Unified IP Phone 7906G – 15 Cisco Unified IP Phone 7931G - 30 Cisco Unified IP Phone 7961G– 150 Cisco Unified IP Phone 7970G - 10 Cisco 7936 IP Conference Station – 4 Cisco IP Communicator (softphones) – 21

• PSTN Gateways One (1) Cisco Integrated Service Routers, Model 2821.

o Two (2) VWIC-2MFT1 – 2-port T1 modules, providing 4 T1 ports.

o One (1) VIC2-2FXO 2-port voice interface card and two (2) VIC2-4FXO 2-port voice interface cards – for a total of 10 FXO ports.

• Analog FXS Gateways One (1) Voice Gateways (VG) 224 to provide 24 FXS ports for

analog phones, modems, fax machines, etc. • Power Failure Transfer Unit

One (1) Gordon Kapes Power Failure Bypass Unit (8 ports) Large Branch Location: The Large Remote Location contains the following Cisco components:

• Cisco 2851 Integrated Services Router with SRST for 96 phones

One (1) VIC2-4FXO, four-port FXO module One (1) VWIC-2MFT-T1, two-port module

• Cisco Unified IP Phones Cisco Unified IP Phone 7906G – 8 Cisco Unified IP Phone 7931G - 10 Cisco Unified IP Phone 7961G - 55 Cisco Unified IP Phone 7970G - 2 Cisco 7936 IP Conference Station – 2 Cisco IP Communicator (softphones) – 10



• Analog FXS Gateways One Voice Gateways (VG) 224 to provide 24 FXS ports for

analog phones, modems, fax machines, etc. • Power Failure Transfer Unit


Medium Branch Location: The Medium Remote Location contains the following Cisco components:


One (1) VIC2-2FXO, four-port FXO module and two (2) VIC2-4FXO, four-port FXO module for a total of 10 FXO ports

• Cisco Unified IP Phones Cisco Unified IP Phone 7906G – 3 Cisco Unified IP Phone 7931G - 8 Cisco Unified IP Phone 7961G – 25 Cisco Unified IP Phone 7970 - 1 Cisco 7936 IP Conference Station – 2 Cisco IP Communicator (softphones) – 5 Cisco ATA 186 2-port Adapter (FXS) analog phones, modems,

fax machines, etc. - 3 • Power Failure Transfer Unit

One (1) Gordon Kapes Power Failure Bypass Unit (8 ports) per satellite office

Small Branch Location: The Large Remote Location contains the following Cisco components:


One (1) VIC2-2FXO, two-port FXO module and one (1) VIC2-4FXO, four-port FXO module for a total of 6 FXO ports

• Cisco Unified IP Phones Cisco Unified IP Phone 7931G - 1 Cisco Unified IP Phone 7961G – 7 Cisco ATA 186 2-port Adapter (FXS) analog phones, modems,

fax machines, etc. - 1 • Power Failure Transfer Unit


1.3.0 System Design Platform The proposed system solution may be based on either of the following two architecture system design:

• Converged TDM/IP: call telephony server supporting LAN/WAN distributed circuit switched port interface cabinets with equipped media gateway interfaces for IP port connectivity

• Client/server: call telephony server supporting media gateway equipment (server-embedded, standalone, switch/router-equipped or



desktop) for non-IP port connectivity Vendor Response Requirement: Briefly and clearly describe the architecture and design elements of the proposed IPTS solution. Include in your basic system description information about the following common equipment hardware elements:

1. Type of architecture design (converged or client/server) 2. Call telephony server and associated common control equipment 3. If applicable, circuit switched port interface equipment housing TDM

port interface circuit cards and media gateway boards. 4. If applicable, LAN-connected media gateways (server-embedded,

standalone, switch/router-equipped, desktop

Cisco Response: Cisco is proposing a distributed IP communications system based on Cisco’s IP Communications architecture. Call processing and applications servers are centralized. Gateways, IP and analog phones, and other resources such as SRST optional software are distributed across VoiceCon’s Headquarters, Regional office and remote locations.

1.3.1 Common Control The primary common control complex of the proposed IPTS should be based on a standalone call telephony server or a call processor blade that is embedded in common equipment that functions as a call telephony server. The physical equipment may either be a fully bundled proprietary hardware/software offering that is factory configured or third party equipment provided by VoiceCon that is capable of running proposed proprietary call processing software without any service degradation. Any and all of the proposed primary common control call processor elements used to provide call processing functions must be proposed in a redundant duplicated design with seamless switchover operation between active and standby control elements, i.e., all active call connections must remain up during switchover in case of failure or major alarm states and new calls set-up without delay. The secondary standby control element must be local to the primary, i.e., physically located at the same VoiceCon facility. The secondary standby cannot be located at a remote VoiceCon facility. This takes into consideration the possibility of simultaneous primary control and LAN/WAN link failure that affects telecommunications services to station subscribers. This redundancy requirement does not apply to local survivable processors at SB facilities where primary control is located at the HQ facility. Solutions based on fully dispersed call processing system designs, i.e., primary control elements at HQ, RO, and SB facilities, however, must conform to the local redundancy requirement wherever a primary control element is installed.



The overall common control design may be based on a load sharing design in which any call telephony server/processor blade may be programmed to function in primary and secondary backup modes. All primary common control elements must be capable of supporting required equipped and wired capacities at time of installation. The call processing rating for the proposed primary and secondary IPTS call server(s) or equivalent(s) must minimally support the following call processing ratings at each of the following VoiceCon facilities: HQ: 35,000 Busy Hour Call Completions (BHCCs) RO: 10,000 BHCCs SB1: 5,000 BHCCs SB2: 2,500 BHCCs SB3: 1,000 BHCCs Vendor Response Requirement: Provide a brief description of the common control design you are proposing in terms of design platform: call telephony server, call processor blade (including necessary housing), third party server. Confirm that the duplicated common control requirement is fully satisfied by the proposed solution, and identify any feature/function that is not available if a standby (back-up) call processing element must be activated in case of a primary element failure.

Cisco Response: Four Cisco Media Convergence Servers (MCS), model MCS 7835-H2, are included in the configuration as load sharing, redundant Unified CallManager call processing servers configured as a single CallManager cluster. Unified CallManager software and all configuration databases are fully duplicated and mirrored on the four redundant servers. These servers are rated at 50,000 BHCC’s per server (125,000 BHCC’s per cluster). The cluster is divided across two locations, the headquarters and regional office, to provide clustering over the WAN. Two servers are located the headquarters (active and backup) and two servers are located at the regional office (active and backup). This configuration ensures that each site can back up the other sites across the WAN link. If the WAN link fails between headquarters and the regional office, each of these sites can operate independently with local redundancy provided by the two servers. It should be noted that the servers are sized such that only one server is required to support all the devices on the system. The MCS 7835-H2 Unified CallManager servers and the MCS 7845-H2 Unity server also include redundant hot-swap power supplies, hot-swap RAID hard drives, redundant ROM, redundant hot-swap fans, and a dual port gigabit Ethernet controller. The equivalent functions of DTMF registers/senders, conference bridges, etc., are part of the Unified CallManager software and therefore are also fully duplicated.



Cisco Unity Unified Messaging is configured on one MCS 7845-H2 server. If a switchover is required between Unified CallManager servers, calls in progress are preserved. The customer configuration database is mirrored on the four servers so customer programmed features are not affected. There is no loss of feature function or capability, or loss of feature transparency due to switching from active to backup servers.

1.3.2 CPU Make/Model Vendor Response Requirement: Identify the make/model of all proposed common control CPU(s) and associated BHCC rating for the configured system.

Cisco Response: The MCS 7835-H2 Unified CallManager Appliance has been proposed for Cisco Unified CallManager; the MCS 7845-H2 has been proposed for Cisco Unity Unified Messaging. These two servers are described in more detail below:

MCS 7835-H2 The Cisco MCS 7835-H2 Unified CallManager Appliance features an Intel Woodcrest dual-core Xeon 2.33GHz processor, an 1333-MHz front side bus (FSB), 2GB SDRAM and two 72GB SAS hard disks. It provides high reliability through the use of RAID 1 disk mirroring and by providing redundant power supplies. Remote management is also provided through support of HP iLO 2 with both iLO Standard and iLO Advanced licensing provided at no additional cost. The Cisco® MCS 7835-H2 Unified CallManager Appliance (MCS 7835-H2) is a high-availability server platform for Cisco Unified CallManager 5.1 and greater and an integral part of a complete, scalable architecture for a new generation of high-quality Unified Communications solutions for enterprise data networks. Delivering the high performance and availability that today’s enterprise networks demand, this solution is easy to deploy and highly cost-effective. The server appliance is preinstalled with an operating system and Cisco Unified CallManager 5.1 or greater; it is fully operational upon startup, requiring entry of just a few configuration items such as IP address and domain. At just 2 rack units (2RU) high, the Cisco MCS 7835-H2 offers tremendous power in a low-profile chassis that minimizes rack space. The MCS 7835-H2 is rated at 50,000 BHCC per server and 125,000 BHCC per cluster. It supports up to 2500 IP phones per appliance and up to 10,000 IP phones in a cluster configuration. MCS 7845-H2 The Cisco MCS 7845-H2 occupies only 2 mounting spaces on a standard third party rack and features dual Intel 5140 dual-core Xeon 2.33GHz processors, 4GB SDRAM and four 72GB SAS hard disks. It provides high reliability through the use of RAID 1 disk mirroring and by providing redundant power supplies. Remote management is also



provided through support of HP iLO 2 with both iLO Standard and iLO Advanced licensing provided at no additional cost.

1.3.3 Call Processing O/S Vendor Response Requirement: Identify the primary operating system of the common control call processor. A version of Linux is preferred, but not mandatory.

Cisco Response: Linux for Cisco Call Manager 5.1; Microsoft Windows 2000 for Cisco Unity.

1.3.4 Memory Vendor Response Requirement: Briefly describe the main memory design and storage elements and capacity for both the generic software and customer database as proposed.

Cisco Response: The MCS 7835-H2 servers have 2 GB PC2-5300 667-MHz double data rate 2 (DDR2) memory with online spare capabilities

1.3.4.1 Database Integrity Vendor Response Requirement: How does the proposed IPTS solution maintain the integrity of the customer database between back-ups?

Cisco Response: The Cisco IPT solution uses a readily available 3rd party vendor database for the data warehousing. As such, the Cisco IPT solution relies on the vendor’s data integrity mechanisms to ensure the integrity of the Cisco inserted data. Cisco uses a well defined data dictionary, rule set and validation for any data insertion/modification/removal of elements from the database. This rule set will ensure no data will be orphaned in the database tables. Multiple administrative changes to the same object will be completed in serial operations and not parallel operations. The database also maintains a transactional log such that any servers that are offline during an update will automatically receive the update when they come back on line.

1.3.4.2 Database Information Loss Vendor Response Requirement: Identify under what circumstances can customer database information (configuration, messages, logs, etc.) be lost during back-ups

Cisco Response: During database backup, no data can be lost. When the database backup operation starts, the backup application changes the database into a read only database. Once the database backup is complete,



the database will be changed back into read/write mode. Changes to the database will be disallowed during this time. If the optional CDR database is backed up at the same time, the system will stop the CDR insert operations until the database is backed up, so no CDR records will be lost during the backup operation.

1.3.4.3 Database Backup Scheduling Vendor Response Requirement: How often should the customer database be backed up? Specify if it is a full or incremental backup and the time the process takes.

Cisco Response: The Cisco IPTS provides a backup utility that will backup and store all relevant databases and files of the IPTS installation. The backup utility allows for both incremental and complete backup. The backup scheduled is recommended for incremental every weekday and Saturday night and full backups every Sunday night. The amount of time to complete the backup will depends on the size of the database and if the backup will be backing up additional servers. For a backup of 1600 user system without CDR records will take approx 10 minutes. Compression and offloading of the data (running as a low priority process) will take an additional 30 minute and would generally be scheduled to run in off hours.

1.3.4.4 Data Purging/Archiving Vendor Response Requirement: Describe the mechanism for data purging and archival, including storage and retrieval of archived data.

Cisco Response: CDR data is stored on the primary server in the cluster. CDR records can be purged based on size or age. There is also an option to archive the CDR’s during the backup procedure. The restoration of CDR data can be done via the Backup utility that is part of the server. Trace file storage is also administratively controllable. The quantity, size and generation of trace files can be configured through the administrative interface. There are additional tools that will allow for the automated capture and storage of trace files from a server or cluster. This tool can capture traces from all servers or specific servers, zip the files and archive them for review.

1.3.5 Power Supply Vendor Response Requirement: Briefly describe common control power requirements and the integrated power distribution design. Indicate if the power supply is dependent on either an AC or DC current source.

Cisco Response: The MCS 7835-H2 servers in this proposal include



redundant hot swap 575 Watt power supplies (AC). Optional DC power supplies are available for power installations backed up by battery.

1.3.5.1 Power Safeguards Vendor Response Requirement: Describe any power failure safeguards that are included in the IPTS design. Briefly describe what happens to system operation during a power failure

Cisco Response: As proposed, this system does not include a UPS or battery backup system. Those can be added optionally once VoiceCon determines their overall requirements for power failure backup. As configured then, in the event of a power failure, the system would shut down. It would automatically restart following restoration of power. This restart would take approximately three minutes.

1.3.5.2 Power Backup Vendor Response Requirement: Is the proposed IPTS solution equipped with standard UPS hardware, and if so how long can the system run on it? If not, what UPS requirements are recommended?

Cisco Response: UPS would be an option and is highly recommended. However, in order to size a UPS system Cisco would need to know the exact configuration of the underlying data infrastructure as well as the duration of backup time needed by VoiceCon. Once these variables are known, Cisco partners such as, APC or TrippLite could specify a properly sized UPS system (or multiple distributed systems in this case).

1.3.6 Ethernet Call Control Signaling Links Vendor Response Requirement: Identify for each active and standby call telephony server the number of available and configured RJ-45 Ethernet LAN uplink interfaces for call control signaling to LAN-connected cabinets/carriers and/or standalone ports. Include a brief description of how the physical Ethernet connection is provided: dedicated circuit board; daughterboard; fully integrated RJ-45 connector, et al.

Cisco Response: The MCS 7835-H2 has a dual onboard 10/100/1000 Network Interface Card (NIC) with two RJ-45 connectors on rear of server. 100BASE-TX cable support can be provided with Category 5 UTP (2 pair) up to 328 feet (100 meters). 1000BASE-T cable support requires Category 5 UTP, 5E UTP, 6 UTP (2 pair) up to 328 feet (100 meters).

The two NIC ports have the same IP address and would be connected to two separate switches for the highest availability.

1.3.7 System Clocks



Vendor Response Requirement: Identify the number and type of internal system clocks that are available and configured.

Cisco Response: All Cisco products including Unified CallManager, Unity, gateways and phones provide internal clocks or can derive clock from a Network Time Protocol (NTP) source. The phones can also derive clock from Unified CallManager. So in most customer deployments the servers and the gateways derive clock from NTP, and the phones derive clock from Unified CallManager. On the digital TDM interfaces (i.e. PSTN-facing T1/E1 circuits or internally-facing T1/E1 circuits to downstream gateways or other digital equipment), the gateways support providing clock to the circuit or deriving clock from the circuit(s) and propagating that clock to other [downstream] circuit(s).

1.3.8 Redundant system design elements It is desirable to have a highly redundant system design, especially as it relates to common control elements necessary for call processing, maintenance, and administration operations. Vendor Response Requirement: Specify the level or degree of redundancy included in your proposal for each of the following listed common control elements. For example, full duplicated back-up, standby load sharing, seamless switchover, cold standby, et al.

• Primary call processor • Main system memory • Customer database memory • RJ-45 Ethernet uplinks to network • Power supply • Tone generators • Call classifiers • Registers • DTMF receivers • I/O interfaces

Cisco Response: Four MCS 7835-H2 Unified CallManager servers are included in the configuration as load sharing, redundant call processing servers. Unified CallManager software and all configuration databases are fully duplicated and mirrored on the four redundant servers. Cisco Unity Voice Mail is configured on a single MCS-7845-H2 server. The MCS 7835-H2 Unified CallManager servers and the MCS 7845-H2 Unity server include redundant power supplies, RAID hard drives, redundant fans, redundant ROM and dual NIC ports. The equivalent functions of DTMF registers/senders, conference bridges, etc., are part of the Unified CallManager software and therefore are also fully duplicated.



1.4 Local Survivability It is important to VoiceCon that station users at all network locations have access to telephony services at all times. This includes 100% of generic software features and trunk circuit access to a local exchange carrier. For this reason it is highly desirable that station users at VoiceCon’s SB facilities have access to telephony services in case of HQ-SB WAN link failure due to switch, router, or private network transmission service issues, or HQ common control failure for any reason. It is preferable that standby telephony services be provided by an on-site call processing option. A less desirable, but acceptable, emergency option is an alternative PSTN-based call control signaling link, but only if an on-site call processing option is not available as part of the system solution. It is also highly desirable that the standby call processing option provide stations users with the same level of telephony services, i.e., station, attendant, and system features, supported by the HQ IPTS at the medium (50 stations) and large (100 stations) SB facilities. For the small (10 stations) SB facility it is acceptable that POTS-like survivability (dial tone, PSTN trunk access, intercom calls, basic features such as Hold and Transfer) is supported. Please note that Power Failure Transfer Station (PFTS) is not acceptable as the local survivability option at the small SB facility. SB facility local survivability for any disruption due to any circumstance (common control failure and/or LAN/WAN incidents) of HQ-based IPTS call control signaling is a mandatory requirement for proposal submission. Vendor Response Requirement: Describe the proposed local survivability solution that satisfies the stated requirements. Include a description of all proposed and priced local survivability options (including any and all required hardware, software, and PSTN transmission services necessary to implement the option) for each of the three SB facilities: 10 stations, 50 stations, 100 stations.

Cisco Response: Cisco offers enterprises of all sizes a cost effective, reliable solution for providing continuous IP Telephony services to branch offices using Cisco Unified Survivable Remote Site Telephony (SRST). A unique, industry-first capability embedded in Cisco IOS Software running on Cisco routers, Cisco Unified SRST provides feature-rich call processing redundancy for centralized Unified CallManager deployments, while leveraging the existing network infrastructure at the remote office. If the WAN link to the remote office fails and the connection to the Cisco Unified CallManager is lost, the branch office phones are automatically redirected to the Cisco Unified SRST branch router, which takes over and provides a core/critical subset of the functions provided by Cisco Unified CallManager -- minimizing the impact to the business. Once the disrupted WAN link is restored, the phones automatically reregister with the original Cisco Unified CallManager -- no manual intervention is required. Cisco Unified SRST is accomplished through



an integrated system approach with no additional hardware components and is currently available for Cisco 175x, 1760, 2600, 2800, 3700, 3800 and 7200 Series Routers. SRST supports up to 24 phones on the 175X series routers, and up to 720 phones on the 3845 series routers. To meet this requirement a Cisco ISR router has been included in the proposal at each remote site which performs both the gateway function as well as acting as the backup local cal processing agent using Cisco Unified SRST. The following diagram illustration the operation of Cisco Unified SRST:

Cisco Unified SRST: Cisco Unified SRST is a feature set of Cisco IOS® Software that runs on the local branch office IP telephony enabled router. Cisco Unified IP Phones and analog gateways maintain a list of call control servers with last resort being the Cisco Unified SRST enabled router. Upon WAN failure the Cisco Unified IP Phone or gateway automatically registers to the Cisco Unified SRST-enabled router which then provides telephony services for locally connected phones and PSTN voice modules. During the WAN failure users are able to receive and place calls both locally and to the PSTN. In addition, a message appears on all display phones advising the remote office of the WAN failure. Upon restoration of the WAN connectivity, the system automatically shifts call-processing function back to the centralized Cisco Unified CallManager cluster. Configuration for Cisco Unified SRST is only needed at install at the central site with all adds, moves and changes being handled at the centralized Cisco Unified CallManager cluster. With no administration required at the remote sites, costs for deploying and maintaining IP Communications to remote offices are greatly reduced providing cost savings to the organization. Cisco



Unified SRST provides a cost effective solution with a set of telephony features provided for the duration of the failure including access to the Public Switched Telephone Network (PSTN) that is not available from other traditional telephony solutions. Cisco Unified SRST also supports secure voice calls during the WAN failure. The Cisco Secure SRST feature allows Cisco Unified IP Phones using secure voice beginning with Cisco Unified CallManager 4.1 and Unified SRST version 3.3, to continue to make secure calls with authentication and encryption for both signaling and media transmission. Cisco Unified SRST supports SIP Phone failover as well. The SIP SRST feature on the Cisco Unified SRST router provides backup SIP Proxy/Redirector for SIP-based IP Phone fallback to the SIP SRST router when network connectivity to the central Unified CallManager(s) is down. This feature provides local SIP call processing on the Cisco Unified SRST router during WAN outage, thus provides back up for Telephony services including off net calls to 911 services etc, and data connectivity can be provided via dial on demand routing (DDR). The Cisco Unified SIP Phones or third party SIP phones will attempt to REGISTER with SRST when connectivity to the central Unified CallManager(s) is lost, and will automatically re-home to the central Unified CallManagers as soon as they become available again (with some hold timers built in to avoid re-homing to quickly in the event of a link-flapping condition). The following features are supported when in SIP SRST mode:

• Local IP Phone-to- IP Phone calls • Local IP Phones-to-PSTN calls (DOD or two-stage dialing), PSTN-to-

local IP Phones (DID or two-stage dialing) • Class of Restrictions (COR) assignable to local IP Phones for outgoing

PSTN calls (e.g. to block 1-900 calls) • Extensive dialplan mapping/manipulation to provide for seamless use

of internal dialing (e.g. 5-digit dialing converted to fully-qualified PSTN dial plan) so that users can continue to use the normal dial plan even though the calls are being diverted to the PSTN.

This features supports Cisco SIP Phones and 3rd-party RFC3261-compliant IP Phones as well as Cisco SIP analog and digital gateways

1.4.1 Survivable IPTS Features/Services Vendor Response Requirements: Identify any required generic software feature (See Section 5.0 Call Processing Features) not available or operational when the local survivability solution is activated at VoiceCon’s 50 station and 100 station SB facilities. Also identify any type of station user equipment (instruments, consoles, softphones, wireless communications devices, et al) not supported in standby survivability mode at these two facilities.

Cisco Response: Key features available during SRST operation include; Placing and receiving calls, call hold, last number redial, speed dial, Do not



Disturb, multi-line appearance, shared lines, alias, class of restriction (COR), and Cisco Unity Express support. All Cisco Unified IP Phones are supported, plus for analog devices the VG248, VG224 or ATA gateway can be used. PSTN module support includes analog and digital trunks including FXO, DID, T-1/E-1, BRI and PRI. SRST (Current Version 4.0) supports the following features: Features:

• Support for all models of Cisco Unified IP Phones, including wireless, and basic-telephone-service (analog POTS) phones

• Re-homing of Unified IP Phones and SIP phones upon failure to branch router for call processing

• Call preservation of local extension-to-extension calls upon failure, and maintenance of existing calls upon recovery (with no time-out)

• Call preservation for extension-to-public switched telephone network (PSTN) calls upon failure (with no time-ou)

• Up to six lines per phone • Call hold and pick up • Speed and last-number redial • Primary line support • Calling-party name • Caller ID and asynchronous-network-interface (ANI) support • Class of restriction • Music on hold (MOH), tone on hold, and music and tone on transfer

(MOH for endpoints PSTN only) • Distinctive ringing • Transfer to voicE-Mail pilot number using PSTN • Transfer across H.323 network of Cisco endpoints • Dual line appearance per button • Three-party G711 temporary conferencing • Call transfer with consult • MOH multicast and live-feed support • Cisco Unified CallManager phone language support • Global call-forwarding enhancement • In-band dual tone multifrequency (DTMF) voicE-Mail integration • Enhanced dial-plan pattern

Facilities Supported:

• WAN link support: Frame Relay, ATM, Multilink Point-to-Point Protocol (MLPPP), serial, ATM Adaption Layer 2 (AAL2), and DSL

• Direct inward dialing (DID) and Direct Outward (DOD) Dialing • Analog foreign exchange office (FXO) and foreign exchange station

(FXS) • PSTN T1 and E1 channel-associated-signaling (CAS) trunks support • ISDN Basic Rate Interface (BRI) and Primary Rate Interface (PRI)

support • E1 R2 signaling support • Secondary dial tone



• Symmetric Session Initiation Protocol (SIP) gateway-to-gateway DTMF relay

• RFC 2833 DTMF relay support

Miscellaneous Features: • Configurable system message • Improved debugs for phones • Call-detail recording and RADIUS server • Inter-working with Cisco Gatekeeper • Alias lists for unregistered phones • Translation rules support • Tool Command Language (TCL)-based simple AA and interactive

voice response (IVR) on local gateways • Secure calls (i.e. authenticated and encrypted signaling and media)

between Unified IP Phones and between Unified IP Phones and the SRST router providing PSTN gateway access. Note: beginning with Unified CallManager release 4.1(2) and SRST release 3.3.

1.4.2 Local Survivability Failover and Switchback Vendor Response Requirements: For each of the SB facilities is failover to the local survivable call processing option seamless, i.e. no interruption of in-process telephony services, for any or all stations users if WAN connectivity is disrupted to the HQ IPTS? Indicate in answer if there is delay for implementing new calls immediately after the WAN disruption. Also describe the switchback process when HQ facility IPTS call control is again available via the WAN, specifying if the process is automatic or manual and how long the process takes to implement. Are connected calls and voice operations at the remote facility affected in any way by the switchback process and how soon can new calls be implemented?

Cisco Response: In the event the Unified IP Phones are unable to reach their central Unified CallManager servers, whether because of a wide area network (WAN) failure or a failure of the Unified CallManager servers themselves, Cisco Unified SRST automatically detects the failure in the network and, using Cisco Simple Network Automated Provisioning (SNAP), initiates a process to intelligently auto configure the router to provide call-processing backup redundancy for the Cisco Unified IP Phones in that office. The local router takes over call processing and telephony features, which stay available without a hiccup to branch employees through the duration of the failure. The only calls lost would be calls on the WAN at the time of the actual failure. Upon WAN failure the Cisco Unified IP Phones and gateways automatically register to the Cisco Unified SRST enabled router which then provides telephony services for locally connected phones and PSTN voice modules. During the WAN failure users are able to receive and place calls both locally and to the PSTN. In addition, a message appears on the phones advising the remote office users of the WAN failure. Upon restoration of the WAN connectivity, the system automatically shifts all call processing and telephony features back to the centralized Cisco Unified CallManager cluster.



How long does it take Unified IP Phones to fallback to the SRST router? It depends on the failure situation and which protocol the phones and gateways are using. You get an "instant" fail-over if and only if:

• The phones are using TCP for signaling and have a hot standby TCP socket already open to the SRST device (note: some Unified IP Phones use UDP for signaling, in which case the timers are a bit different/longer).

• The TCP connection is explicitly closed by a TCP FIN or RST, so that the phone isn't waiting until it times out.

Otherwise, a [configurable] timeout will occur when waiting for a reply from the server (or upon receiving an ICMP unreachable message) and the phone will initiate the failover after a [configurable] number of attempts.

1.4.3 Survivable Messaging Services It is desirable that remote station users at the RO and SB facilities have access to messaging services if there is a WAN link disruption to the HQ messaging system. Vendor Response Requirements Does the proposed IPTS network and messaging solution satisfy this requirement if WAN connectivity between HQ and any of the other facilities (RO, SBs) is not available? Briefly describe how messaging services would be implemented and accessed by remote station users in emergency situations. The minimum messaging services function in survivable mode should include voice mailbox access by station users.

Cisco Response: In the event of a WAN failure, the direct connection between the remote phone and Unity is lost. Users can receive and retrieve Voice Mail messages, but the messaging waiting indicator (MWI) would be disabled during the failure. Users may rely on the "Missed calls" shown on the phone display to check for their voice mail. Voice Mail access is accomplished by configuring the Cisco Unified SRST router with the number to call when users press the “Messages” button on their Unified IP Phones. For example, let’s assume that normally the users dial a 5-digit extension to access the voicemail pilot. When in SRST mode, the SRST router takes that 5-digit number and inflates it to the fully-qualified E.164 address so that the call to the voicemail pilot number can be routed via the PSTN. Voice Mail can be received using the support for voicemail integration with Unity server via Analog/DTMF. Cisco Unified SRST forwards in-band analog/DTMF messages to the central Unity voicemail server so that call forward busy/no answer/all to the Unity server can be properly routed via the PSTN. This means that personal greetings can be heard on direct access calls and when calls are forwarded to voicemail (note that FXO hairpin forwarded calls to voicemail must have dis-connect supervision from the central office).



1.4.4 Network Failover Resiliency In the unlikely event the redundant common control complex (primary active and secondary backup) at either HQ or RO facilities become nonfunctional due to extreme system failure or catastrophic circumstances, e.g., fire, VoiceCon requires implementation of a resilient network failover process. This process requires that all local station users and media gateway equipment configured behind the nonworking common control complex automatically re-register to designated emergency call telephony server(s) at either the local or remote facility for continuity of telephony services. For this reason it is necessary that the designated emergency call telephony server(s) located at the HQ/ RO facility be capable of supporting sufficient port capacity requirements in the event of a failover. Vendor Response Requirements Does the proposed IPTS solution support network failover resiliency in case of a catastrophic common control failure at either the HQ or RO facilities? If affirmative, describe the failover process, optional hardware/software and/or WAN transmission requirements to implement, and the time required for the network failover to be implemented before telephony services are available. Indicate if the proposed IPTS solution can support more than one network failover design.

Cisco Response: A Cisco Unified CallManager cluster can be deployed across multiple sites that are connected by an IP WAN with QoS features enabled. As shown in the diagram in Section 1.0.0 above, the proposed Cisco CallManager cluster has been split across two sites: the headquarters and regional office location. This configuration provides not only the required single system image, but also local and network failover redundancy.

Remote failover allows you to deploy the backup servers over the WAN. Using this deployment model, you may have up to eight sites with Cisco Unified CallManager subscribers being backed up by Cisco Unified CallManager subscribers at another site.

The key advantages of clustering over the WAN are:

• Single point of administration for users for all sites within the cluster

• Network resilience for the entire cluster

• Feature transparency across all users on the cluster

• Shared line appearances

• Extension mobility within the cluster

• Unified dial plan



These features make this solution ideal as a disaster recovery plan for business continuance sites or as a single solution for up to eight small or medium sites.

WAN Considerations For clustering over the WAN to be successful, you must carefully plan, design, and implement various characteristics of the WAN itself. The Intra-Cluster Communication Signaling (ICCS) between Cisco Unified CallManager servers consists of many traffic types. The ICCS traffic types are classified as either priority or best-effort. Priority ICCS traffic is marked with IP Precedence 3 (DSCP 24 or PHB CS3). Best-effort ICCS traffic is marked with IP Precedence 0 (DSCP 0 or PHB BE). The following design guidelines apply to the indicated WAN characteristics:

Delay - The maximum one-way delay between any Cisco Unified CallManager servers for all priority ICCS traffic should not exceed 20 ms, or 40 ms round-trip time (RTT). Delay for other ICCS traffic should be kept reasonable to provide timely database and directory access. Propagation delay between two sites introduces 6 microseconds per kilometer without any other network delays being considered. This equates to a theoretical maximum distance of approximately 3000 km for 20 ms delay, or approximately 1860 miles. These distances are provided only as relative guidelines and in reality will be shorter due to other delay incurred within the network. Nevertheless, backup sites can be located a significantly distant locations for maximum benefit.

Jitter - Jitter is the varying delay that packets incur through the network due to processing, queue, buffer, congestion, or path variation delay. Jitter for the IP Precedence 3 ICCS traffic must be minimized using Quality of Service (QoS) features.

Packet loss and errors - The network should be engineered to provide sufficient prioritized bandwidth for all ICCS traffic, especially the priority ICCS traffic. Standard QoS mechanisms must be implemented to avoid congestion and packet loss. If packets are lost due to line errors or other "real world" conditions, the ICCS packet will be retransmitted because it uses the TCP protocol for reliable transmission. The retransmission might result in a call being delayed during setup, disconnect (teardown), or other supplementary services during the call. Some packet loss conditions could result in a lost call, but this scenario should be no more likely than errors occurring on a T1 or E1, which affect calls via a trunk to the PSTN/ISDN.

Bandwidth - Provision the correct amount of bandwidth between each server for the expected call volume, type of devices, and number of devices. This bandwidth is in addition to any other bandwidth for other applications sharing the network, including voice and video traffic between the sites. The bandwidth provisioned must have QoS enabled to provide the prioritization and scheduling for the different classes of traffic. The general rule of thumb for bandwidth is to over-provision and under-subscribe.



Quality of Service - The network infrastructure relies on QoS engineering to provide consistent and predictable end-to-end levels of service for traffic. Neither QoS nor bandwidth alone is the solution; rather, QoS-enabled bandwidth must be engineered into the network infrastructure.

Intra-Cluster Communications In general, intra-cluster communications means all traffic between servers. There is also a real-time protocol called Intra-Cluster Communication Signaling (ICCS), which provides the communications with the Cisco CallManager Service process that is at the heart of the call processing in each server or node within the cluster.

The intra-cluster traffic between the servers consists of the following:

• Database traffic from the IBM Informix Dynamic Server (IDS) database that provides the main configuration information. The IDS database is replicated from the publisher server to all other servers in the cluster using best-effort. The IDS traffic may be re-prioritized in line with Cisco QoS recommendations to a higher priority data service (for example, IP Precedence 1 if required by the particular business needs). An example of this is extensive use of Extension Mobility, which relies on IDS database configuration.

• Firewall management traffic, which is used to authenticate the subscribers to the publisher to access the publisher's database. The management traffic flows between all servers in a cluster. The management traffic may be prioritized in line with Cisco QoS recommendations to a higher priority data service (for example, IP Precedence 1 if required by the particular business needs).

• ICCS real-time traffic, which consists of signaling, call admission control, and other information regarding calls as they are initiated and completed. ICCS uses a Transmission Control Protocol (TCP) connection between all servers that have the Cisco CallManager Service enabled. The connections are a full mesh between these servers. Because only eight servers may have the Cisco CallManager Service enabled in a cluster, there may be up to seven connections on each server. This traffic is priority ICCS traffic and is marked dependant on release and service parameter configuration.

• CTI Manager real-time traffic is used for CTI devices involved in calls or for controlling or monitoring other third-party devices on the Cisco Unified CallManager servers. This traffic is marked as priority ICCS traffic and exists between the Cisco Unified CallManager server with the CTI Manager and the Cisco Unified CallManager server with the CTI device.

Failover Between Subscriber Servers With Cisco Unified CallManager Release 5.x, the device configuration records are cached during the initialization or boot-up time. The effect is that Cisco Unified CallManager might take a longer time to initialize, but any failover or failback for all devices is not affected by the delay in accessing the publisher database.



1.5 Session Initiated Protocol (SIP) VoiceCon requires that the proposed IPTS support SIP-compatible stations and trunk networking as specified in the most current IETF Work Group RFC document. It is also required that the IPTS solution be capable of supporting the IETF-sponsored signaling protocols used for Internet conferencing, telephony services and features, presence, events notification and instant messaging. 1.5.1 SIP Stations Vendor Response Requirements Indicate if the IPTS solution as proposed can currently support SIP-compatible desktop telephone instruments (self and/or third party) and PC client softphones assuming 20% of individual system end user stations are IP-based. Specify if SIP call control is embedded in the IPTS common control design or optional hardware/software elements are required. Also identify up to three (3) third party SIP telephones you have successfully tested for operation behind your proposed IPTS solution.

Cisco Response: Cisco Unified CallManager Release 5.0 supports IETF standard Unified IP Phones. These include Cisco Unified IP Phones with SIP firmware loads as well as third-party SIP Phones. Cisco Unified CallManager supports RFC 3261 compliant devices, so as long as the third party phones are compliant they should work with Unified CallManager Release 5.0. (Appendix A shows a detailed list of the SIP RFC’s that are supported.) SIP call control is native to Unified CallManager 5.0 (i.e., embedded) and does not require additional hardware or software components. Note that Unified CallManager 5.0 also simultaneously supports all of the other existing call control protocols including SCCP, MGCP, H.323, Q.SIG, CTIQBE and others, and provides rich inter-working between them. Third-party phones have specific local features that are independent of the call control signaling protocol, such as features access buttons (fixed or variable). Basic SIP RFC support allows for certain desktop features to be the same as Cisco Unified IP Phones and also allows for interoperability of certain features. However, these third-party SIP phones do not provide the full feature functionality of Cisco Unified IP Phones.

Cisco is working with key third-party vendors who are part of the Cisco Technology Development Partner Program and who are developing solutions that leverage the new Cisco Unified CallManager and Cisco Unified CallManager Express SIP capabilities.

Cisco is also participating in an independent third party testing and interoperability verification process being offered by tekVizion. This independent service provided by tekVizion has been established to enable third-party vendors to test and verify the interoperability of their endpoints with Cisco Unified CallManager and Unified CallManager Express.



For more information on Cisco's line-side SIP interoperability and third-party verification, visit http://www.cisco.com.

1.5.1.1 SIP Clients Vendor Response Requirements Do the IP desktop telephone instruments and PC client softphones included as part of this proposal in response to RFP Section 5: Voice Terminals currently conform to IETF SIP standards? If not, are they upgradeable to support SIP standards and specifications via a firmware download if required in the future? If a firmware download is required is there an associated cost or fee to VoiceCon?

Cisco Response: The Cisco Unified IP Phones included in this proposal support IETF Standard SIP. They can be ordered as Unified IP Phones, or ordered as SCCP (Cisco Skinny protocol) phones and later field upgraded with SIP firmware loads if desired. The Cisco 7936 Conference Station and the Cisco IP Communicator softphone included in this response are only available as SCCP endpoints at this time. The Cisco IP Communicator softphone will support SIP in a future release. The 7936 Conference Station however will not.

1.5.2 SIP Trunk Networking Vendor Response Requirements Indicate if the IPTS network solution as proposed can support SIP-based trunk networking. Specify if SIP media proxies are required to support this requirement. Identify up to three (3) major Service Providers (SPs) and three (3) other IPTS suppliers you have conducted IP trunk networking compatibility tests with for the proposed IPTS.

Cisco Response: Cisco has supported SIP-based trunking since Unified CallManager Release 4.0. SIP call control is native to Unified CallManager 5.0 (i.e., embedded) and does not require additional hardware or software components. Note that Unified CallManager 5.0 also simultaneously supports all of the other existing call control protocols including SCCP, MGCP, H.323, Q.SIG, CTIQBE and others, and provides rich inter-working between them. Cisco Unified CallManager trunk connections support both H.323 and SIP. In many cases, the decision to use H.323 or SIP is driven by the unique feature(s) offered by each protocol. There are also a number of external factors that can affect the choice of trunk protocol, such as customer preference or the protocol's maturity and degree of interoperability offered between various vendors' products.

For trunk connections between Cisco devices, this decision is relatively straightforward. For trunk connections to other vendors' products and to service provider networks, it is important to understand which features are required by the customer and the extent of interoperability between any two



vendors' products. Currently, major Tier 1 Service Providers are in the process but have not yet concluded their verfication testing for SIP-based trunking with Cisco Unified CallManager.

Further information about how to configure and use SIP trunks can be found in the CallManager 5.0 System Reference Design Guide posted at:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementation_design_guide_book09186a00806492bb.html

1.5.2.1 SIP Applications Vendor Response Requirements: Indicate if the IPTS network solution as proposed can support SIP-enabled applications, such as Internet conferencing, telephony services and features, presence, events notification and instant messaging.

Cisco Response: In addition to supporting SIP trunks natively in Cisco Unified CallManager for application support, Cisco builds upon that functionality with Cisco Unified Presence Server. In addition to core SIP support, Cisco Unified Presence Server uses SIMPLE (SIP for Instant Messaging and Presence Leveraging Extensions) technology to support both instant messaging (IM) and presence. Cisco offers the Cisco Unified Presence Server 1.0 as an optional application. This option is described in detail below. It has not been priced into this RFP response.

The Cisco® Unified Presence Server is a critical component for delivering the full value of a Cisco Unified Communications environment. It collects information about a user's availability status, such as whether or not you are using a communications device such as a phone at a particular time. It also collects information regarding a user's communications capabilities, such whether Web collaboration or video conferencing is enabled. Using user information captured by the Cisco Unified Presence Server, applications such as Cisco Unified Personal Communicator and Cisco Unified CallManager can improve productivity by helping users connect with colleagues more efficiently through determining the most effective way for collaborative communication.

PRODUCT OVERVIEW Cisco Unified Presence Server helps you deploy Session Initiation Protocol (SIP) technology to support new voice services in your enterprise environment. SIP enhances the voice network by providing a core set of behaviors for session establishment and control that can be applied in a wide array of features and services. In addition to core SIP support, Cisco Unified Presence Server uses SIP for Instant Messaging and Presence



Leveraging Extensions (SIMPLE) technology to support both instant messaging (IM) and presence. Cisco Unified Presence Server consists of a SIP presence engine and a SIP proxy function. The presence engine collects user presence information (such as busy, idle, away, or available status) as well as user capabilities (such as the ability to support voice, video, IM, and Web collaboration) and compiles the data in a repository for each user. This repository is accessed by the applications and features that each user employs. Unique user rules and privacy can be applied by each user to ensure that only authorized applications and users have access to presence information. The SIP proxy function allows for efficient and accurate routing of both presence and general SIP messaging through the enterprise. Cisco Unified Presence Server is strongly integrated with various desktop clients and applications. It helps enable Cisco Unified Personal Communicator, the Cisco enterprise desktop client, to perform numerous functions such as click-to-dial and phone control as well as voice, video, and Web collaboration. In addition, Cisco Unified Presence Server provides a core IM service for Cisco Unified IP Phones connected to Cisco Unified CallManager. Cisco Unified Presence Server also supports interoperability with enterprise desktop applications such as IBM Sametime and Microsoft Office Communicator, enabling these desktop applications to operate in conjunction with Cisco Unified IP phones supported on Cisco Unified CallManager.

KEY FEATURES AND BENEFITS

Standards-Based SIP/SIMPLE Network Interface Cisco Unified Presence Server provides a standards-based peering environment to any SIP- or SIMPLE-enabled applications and networks. In effect, any user status that is maintained in Cisco Unified Presence Server can be requested using the IETF standards for status and presence sharing. These SIP and SIMPLE standards define the accepted messaging to initiate and maintain a status request and to provide appropriate responses. Status information can be collected and distributed by Cisco Unified Presence Server, depending on the needs of the services deployed. This implemented interface is providing a standardized way to interoperate with numerous Cisco partners building value-add services for Cisco Unified Communications customers.

Cisco Unified Personal Communicator Network Interface Cisco Unified Presence Server is required to support the core functions for Cisco Unified Personal Communicator, storing personal data including status and capabilities for users as well as their individual rules and preferences. Cisco Unified Presence Server also helps enable phone control and monitoring for click-to-dial service from Cisco Unified Personal Communicator. A highly secure environment helps ensure the integrity of this personal information ranging from user passwords to network connectivity information to personal contact lists. Note: A full description of Cisco Unified Personal Communicator is located at the end of this section.



IP Phone Messenger Network Interface The Cisco IP Phone Messenger service included with Cisco Unified Presence Server provides Cisco Unified IP phones with an IM client complete with presence-enabled contacts lists. Its real-time collaboration capabilities give phone users who might be away from their PCs a quick way to check on the presence status of colleagues (Figure 1). They can also send and receive short text messages, many of them available in a list of phrases and full sentences to save typing them on the phone keypad (Figure 2). Message recipients can reply to their messages or press the Dial soft key to call back without having to look up or dial the number (Figure 3).

Cisco IP Phone Messenger Contacts List

Selecting a Preconfigured Message

Receiving an Instant Message on the Cisco Unified IP Phone

Cisco IP Phone Messenger client features include: • Support for Cisco Unified IP Phone 7905G, 7911G, 7912G, 7920, 7940G, 7941G, 7941G-GE, 7960G, 7961G, 7961G-GE, 7970G, and 7971G models in Skinny Client Control Protocol (SCCP) mode and Cisco Unified IP Phone



7911G, 7941G, 7941G-GE, 7961G, 7961G-GE, 7970G, and 7971G models in SIP mode

• Manual setting of user status as available, busy, busy but interruptible, do not disturb, out of office, vacation, show as logged off, or unavailable

• Configurable contact list of up to 100 contacts per user that shows presence status based on phone activity and PC activity if using Cisco Unified Personal Communicator

• Ability to receive instant messages displayed on phone text display with audible and visible indication (flashing of the phone message waiting indicator light)

• One-button call back to instant message originator

• Ability to view and clear previously received instant messages

• Preconfigured messages to compose or reply to instant messages

Microsoft Office Communicator Interoperability Cisco Unified Presence Server helps users with the Microsoft desktop environment to interoperate with Cisco Unified IP phones connected to Cisco Unified CallManager. It supports Microsoft Office Communicator click-to-dial and phone monitoring functions with all the associated features. This monitoring is done by enabling Microsoft Office Communicator capabilities in Cisco Unified Presence Server and identifying the users who wish to have Microsoft Office Communicator. (Refer to Microsoft specifications for the full requirements and list of capabilities supported by Microsoft Office Communicator.)

PRODUCT ARCHITECTURE AND FEATURES

Software Structure Cisco Unified Presence Server uses the same platform infrastructure as Cisco Unified CallManager 5.0, following its appliance model principles. Cisco Unified Presence Server Appliance is a single software entity that provides access to administration with a GUI and helps enable initial setup and installation on a command-line interface (CLI) similar to those on other Cisco products. Information retrieval takes place through standard interfaces such as Cisco AVVID Extensible Markup Language (XML) and Simple Object Access Protocol (SOAP). Cisco Unified Presence Server 1.0 is a software product that must be loaded onto a Cisco MCS 7825, 7835, or 7845 Media Convergence Server and run as a standalone application. It cannot be deployed co-resident with any other Cisco application on that hardware.

Serviceability Because it runs on the same software infrastructure and hardware as Cisco Unified CallManager 5.0, Cisco Unified Presence Server takes advantage of the same serviceability features. In addition, Cisco Unified Presence Server has further capabilities in the following areas:

• Alarms-The presence engine and SIP proxy components have alarms for various failure scenarios and unexpected behaviors, and the IP phone messenger service has alarms to identify any abnormalities.



• Performance counters-Objects and counters are defined to support the presence engine and SIP proxy components, tracking metrics such as number of actual subscriptions, notifications, registrations, and instant messages sent through Cisco Unified Presence Server.

• Debug and trace utilities- System Diagnostic Interface (SDI) trace capability is supported to provide the ability to log, debug, and analyze specific trace data.

• Service activation-All Cisco Unified Presence Server components and services can be started, stopped, and restarted. Although core network components such as configuration agents are started automatically upon system boot-up, various feature capabilities such as the presence engine and the SIP proxy require manual activation for startup.

• Monitoring-The Real Time Monitoring Tool (RTMT) provides all alarming as well as trace and debug information.

Security The Security module of Cisco Unified Presence Server 1.0 addresses internal environment security as well as external security between Cisco Unified CallManager, Cisco Unified Personal Communicator, and external applications. Its functions include:

• Platform security-The appliance model defined for the common infrastructure within Cisco Unified Presence Server and Cisco Unified CallManager 5.0 includes Secure Shell (SSH) Protocol and Secure File Transport Protocol (SFTP) for access to the platform, as well as HTTPS for access to management applications.

• Internal application security-Cisco Security Agent supports Cisco Unified Presence Server as part of the base application security.

• IP signaling and transport security-Supported through Transport Layer Security (TLS) and IP Security (IPsec).

Administration Interfaces The following administration functions are supported: • System administrator GUI for provisioning of system data and default end-user data

• Bulk Administration Tool for creation of end users

• End-user GUI for provisioning of end-user service data

SIP and SIMPLE Presence Capabilities The following presence functions and capabilities are supported: • SIMPLE core functions (from RFCs 2778 and 2779)

• Subscribe for presence

• Notify of presence

• Publish of presence

• Watcher information and watcher information template package

• Presence event package

• Registration event package

• Resource list subscription

• Presence data information format and extensions



• Rich presence extensions

SIP Proxy Capabilities The following capabilities for proxy of SIP messages are supported: • SIP core proxy functions (from RFC 3261)

• IM routing

• Method- and event-based routing

• Domain Name System service record capabilities (DNS SRV)

• Asserted identity

• Diversion indication

Localization English is the only language supported at initial release of Cisco Unified Presence Server 1.0. The following languages are planned to be supported within 6 months of the initial release: Bulgarian, Catalan, Chinese (Simplified, Hong Kong, and Taiwan), Croatian, Czech, Danish, Dutch, French (Europe), Finnish, German, Greek, Hungarian, Italian, Japanese, Korean, Norwegian, Polish, Portuguese (Europe, Brazil), Romanian, Russian, Spanish (Europe), Serbian, Slovak, Slovenian, and Swedish. Additional Information: Additional information about product specifications, platforms supported, capacities, etc., can be found at:

http://www.cisco.com/en/US/products/ps6837/index.html Cisco Unified Personal Communicator Product Information: An integral part of the Cisco® Unified Communications family of products, Cisco Unified Personal Communicator is a desktop computer application that helps enable more effective communications. By transparently integrating your most frequently used communications applications and services, Cisco Unified Personal Communicator streamlines the communication experience, helping you work smarter and faster. With Cisco Unified Personal Communicator (Figure 1), you can access voice, video, document sharing, and presence information-all from a single, rich-media interface connected to Cisco Unified CallManager.

Example of the Cisco Unified Personal Communicator with Video



SOLUTION OVERVIEW

Communicate More Effectively Cisco Unified Personal Communicator transparently integrates a wide variety of communications applications and services, connecting you to a rich set of unified communication tools. Using dynamic presence information, you can check the availability of colleagues in real time, reducing "phone tag" and improving productivity. You can easily search existing directories to locate important contacts and initiate communications. Video and Web conferencing enable you to exchange ideas "face to face" and collaborate more effectively with colleagues. You also can view and hear your voice messages quickly and easily. With Cisco Unified Personal Communicator, you can connect to important communication tools virtually anywhere, anytime to enable smarter, more effective communications.

Many workers battle communications overload on a daily basis, and they are forced to use a wide variety of devices and tools to communicate with colleagues, partners, and customers. Each application works differently, with its own set of rules, tools, and directories. Cisco Unified Personal Communicator simplifies the communication experience, helping teams and knowledge workers share information faster and communicate in real time.

Reduce Communication Delays with Colleagues and Decision Makers Cisco Unified Personal Communicator helps you determine if co-workers are available or busy before trying to contact them. This availability information is updated automatically using dynamic information from the Cisco Unified Presence Server. You can see immediately who is online, offline, available, or busy. Customized information, such as "on vacation" or "in a meeting",



can also be shared to let you know why someone is unavailable or busy. Knowing whether contacts are available and how best to communicate with them helps reduce communication delays between workers, thereby enabling faster decision making and enhanced productivity. Cisco Unified Personal Communicator also lets you know which method of contact other users prefer-voice, video, or E-Mail.

Streamline Communications Cisco Unified Personal Communicator provides powerful communications features integrated with your desktop computer, including integrated contact lists, click-to-call, voicemail playback, inbound call notification, and media escalation. By being able to control your communications from a single window, you can communicate more effectively and instantly be more productive: • Find contact information quickly by using Cisco Unified Personal Communicator to search your corporate directory.

• Click-to-call from the client and save time by not having to dial telephone numbers.

• Make calls using the integrated soft phone or use Cisco Unified Personal Communicator to control your Cisco Unified IP phone on Cisco Unified CallManager.

• View recent communication activities so that you can respond faster. See who called you and when. View a list of your voice messages on screen, and click to play or return the call.

• Add communication media on demand. When on a call, you can quickly and easily add video or document sharing to enhance collaboration and meeting effectiveness.

• See a list of all participants on a conference call, eliminating the need for roll calls.

• Receive pop-up notifications of incoming calls. See who is calling and the call type-voice only or video call-before you answer. You can accept the call if you are available or divert the call to voicemail immediately with a simple mouse click.

Increase Productivity and Enhance Collaboration With Cisco Unified Personal Communicator, you can enrich communications beyond the realm of voice calls using video and Web conferencing. Interactive face-to-face communications enhances productivity and the quality of communications, streamlines business decision making, and improves teamwork. By reducing the need for in-person meetings, video conferencing also enables companies to save money on travel expenses and time associated with traveling to meetings. Using Web conferencing, you can collaborate with co-workers virtually anywhere, anytime. Cisco Unified Personal Communicator helps you share documents or presentations with people who are located across the street or on the other side of the globe. By integrating virtual meetings into everyday communications, you can expand your market reach, improve operational effectiveness, and speed decisions.




• Communication integration-Take advantage of a single intuitive interface for voice and video calls, voicemail playback, Web conferencing, and integrated directories.

• Presence-View real-time availability of other Cisco Unified Personal Communicator users.

• Unified contact list-Search your corporate directory from one easy-to-use interface to locate contacts quickly. Simply click to call.

• Media escalation-Add communication methods during a session; for example, you can add video to an existing audio session or add Web conferencing to an existing audio or video session.

• Click-to-call-Dial from the contact list, using either the integrated soft phone or an associated Cisco Unified IP phone.

• Integrated voice and video calling-Exchange ideas face to face with a coordinated video display on the PC screen and audio conversation with the soft phone. Users can place video calls to others using Cisco Unified Personal Communicator, Cisco Unified Video Advantage, or the Cisco Unified IP Phone 7985G, a personal desktop video phone.

• IP phone association-Use Cisco Unified Personal Communicator to control your desktop Cisco Unified IP phone and make or receive calls.

• Conferencing-Create voice or video conferencing sessions by simply merging conversation sessions using the Cisco Unified Personal Communicator intuitive interface. There is no need to call into a separate conference bridge.

• Web conferencing-Launch a Web conferencing session at a moment's notice to share content, such as a presentation, with others.

• Voice messages-Access Cisco Unity® Connection voicemail messages-view, playback, sort, and delete messages-all from the same client application.

Additional Information: Additional information about product specifications, platforms supported, capacities, etc., can be found at: http://www.cisco.com/en/US/products/ps6844/index.html

1.6 Security VoiceCon requires a secure IPTS network solution to optimize system performance and reduce the probability of toll fraud and illegal system access. 1.6.1 Authentication Vendor Response Requirements Briefly describe authentication processes embedded in the proposed IPTS solution to prevent: unauthorized access to common control elements, data



resources; and abuse of telephony services, e.g., toll fraud.

Cisco Response: All administration of Unified CallManager is done via HTTPS, which uses server certificate authentication and client username/password authentication. Access to the Unified CallManager OS is also password protected whether it is accessed locally via the console or remotely. Both the console and web application interface allow for role-based authorization that can limit access to configuration so that an administrator only has access to what they need to perform their job. In addition, access to backend services, including, but not limited to database, time synchronization, directory lookups, and APIs, are also authenticated and can be optionally encrypted using SSL or IPSEC (e.g. LDAP requests to a backend directory can use LDAP over SSL for example.). The end-user can manage portions of their account within Unified CallManager via HTTPS, which uses server certificate authentication and client username/password authentication. Access to voicemail is also protected with passwords via the telephony or web user interface and supports two-factor authentication in the TUI. Voicemail messages can optionally be encrypted so that users cannot forward voicemail messages via email. Support for authentication and encryption of signaling protocols (including SCCP, MGCP, H.323, CTIQBE and SIP) and RTP media is also provided. Signaling is secured via Tunneling Layer Security (SSL) or IPSEC and media is encrypted using AES-128bit encryption Secure RTP (sRTP). Cisco Unified IP Phones also support authentication and verification of integrity for firmware images and configuration files using public-key digital signatures. CallManager 5.0 introduced configuration file encryption using AES-128-CBC (Cipher-Block Chained). Beginning in Unified CallManager 4.0, Cisco supports authenticated and encrypted signaling over TLS or IPSec using an RSA signature, HMAC-SHA-1 authentication tag, and AES-128-CM encryption. The security of Cisco Unified IP Phones begins with signed images. This feature allows an Unified IP Phone to validate that the image it receives is an image generated by Cisco. Once a phone has a signed image, it can only be replaced with an image that has a matching signature. This prevents Trojan-Horse images from being installed in phones and subverting other protection mechanisms. Images are digitally signed using a Cisco CA authorized certificate and then wrapped into a Digital Signature Envelope containing a Certificate Identity, Digest Algorithm and Signature Block. When a Unified IP Phone downloads a new image, it compares the Signer Identity and Signature Hash of that image with the Image Signing Trust Anchor in the existing image. If they don't match, the new image is rejected. Beginning with Unified CallManager 3.3(3), all images are signed by default with no configurable options. In CCM 4.0, the configuration file that gets downloaded via TFTP is signed as well. Every Unified IP Phone, Unified CallManager, gateway or application that participates in the authentication and encryption scheme contains a unique X.509v3 digital certificate. These certificates contain, among other things, the public key of the device and the signature of the Certificate Authority (CA)



that issued the certificate. It is the public key / private key pair and the CA’s certificate that form the trust anchor on which all other secure communications rely. A process called the Certificate Authority Proxy Function (CAPF) is used to load certificates into phones, serving as the broker between the phone and the actual certificate authority. CAPF will install what are called Locally Significant Certificates, or LSCs, implying significance with a locally operated certificate authority. These are in contrast with Manufacturing Installed Certificates, or MICs, which are Cisco-rooted certificates installed into some phone models as part of the manufacturing process. In addition to the certificate, devices have another document that contains a list of devices that it will trust on the network. More specifically, the file contains the certificates of all of the devices on the network that the device will trust. In the case of Cisco Unified IP Phones, that file is called the Certificate Trust List, or CTL file. The CTL file contains the certificates of all of the Unified CallManagers, TFTP servers and CAPF servers in the cluster in which the phone participates. It is the combination of the certificate and the CTL file that the phone uses to establish a trusted, bi-directional relationship between the Unified IP Phone and Unified CallManager. This is done as part of the TLS establishment during the phone’s registration with Unified CallManager. In this way, mutual authentication occurs when phones register with Unified CallManager. TLS is a security transport protocol that can carry a wide variety of signature, authentication and encryption algorithms. Cisco uses RSA signatures, HMAC-SHA-1 authentication tags, and AES-128-CM encryption. Through the RSA signature process, Pre-shared Master Secrets are derived which are used as SALT values in the derivation of all future authentication and encryption keys. It is the exchange of RSA signatures that establishes the trusted identity of the two devices. Integrity is the persistent authentication of every SCCP signaling packet thereafter using HMAC-SHA-1 authentication tags. AES-128 is the encryption algorithm that has been proposed by NIST as the next generation replacement for 3DES. It can use several modes. Cisco uses counter mode, designated as AES-128-CM, for encryption of signaling. A lock icon next to a Unified CallManager’s IP address or DNS name displayed on the phone’s “Settings” menu is an indication to the user or administrator that the signaling between the phone and Unified CallManager is both authenticated and encrypted. Finally, Unified CallManager provides a rich set of Toll fraud counter-measures. These capabilities are integral to the Unified CallManagers dial plan and call control logic. All phones, trunks, gateways, voicemail ports, and CTI applications can be assigned a Class of Service which defines what numbers/prefixes they are permitted to reach and how calls to those numbers should be routed and/or manipulated. With the addition of time-of-day



routing, all of the above devices can be further restricted to certain numbers/prefixes during certain hours of certain days. It’s also possible to restrict the ability of the devices listed above from being able to transfer calls externally or conference external parties without an internal participant. Finally, Forced Authorization Codes and Client Matter Codes can be applied to all or some calls based on number/prefix, which will require a code be sent via DTMF from the end user to authorize the call.

1.6.2 Disruption of Services Vendor Response Requirements Briefly describe any embedded features/functions in the proposed IPTS that will reduce probability of telephony services disruption due to Denial-of-Service (DoS) attacks.

Cisco Response: While the most effective methods for stopping DoS attacks are deployed in the infrastructure, Unified CallManager utilizes a number of tactics to manage DoS exposure. Unified CallManager 5.0 comes pre-installed as a stripped down and hardened version of Linux, which greatly reduces the number of DoS attack vectors and hotfixes/service packs that need to be applied. Cisco also provides a very aggressive turn around for posting hot fixes/service packs in the event an attack is identified. Within the Unified CallManager 5.0 OS, access lists are automatically configured which filter traffic based on source/destination ports and source/destination IP addresses. These access lists act to enclose the communication between Unified CallManager servers and only permit clients to connect on certain ports/sockets. For example, native database access is only provided within/between the Unified CallManager servers, but an API (AXL/SOAP) is provided for administrative access to the database from external devices. Cisco Security Agent also comes pre-installed in the Unified CallManager 5.0 OS. CSA provides protection for day-zero attacks as well as well known DoS attacks. It is provided—at no charge—as part of Unified CallManager. It constantly monitors activity on the network, file system, memory, and running processes to determine abnormal behavior and stop it. This helps protect against worms, viruses, trojans, and blended viruses, which are becoming the most popular form of DoS attacks. In addition, the web application interfaces, signaling, console, and backend services support authentication, so it is difficult to take advantage of resources for application-based DoS attacks, which sometimes cannot be stopped in the network. Finally, if a DoS attack was successful against Unified CallManager, the wide array of redundancies Unified CallManager affords you makes it nearly impossible for an attacker to find a silver bullet. Signaling can be handled by multiple Unified CallManager servers separated across a network, so if one



were to be attacked the others could fill its place. If a Unified CallManager were to be taken down by an attack, remote SRST gateways could take over signaling for branches. Unified CallManager services such as, signaling and administration can be separated onto different physical servers, so a vulnerability in one service wouldn’t take down both servers. However, the best method of stopping DoS attacks is in the network infrastructure. Cisco supports a wide array of products that offer a strong defense against attackers trying to deny service to valuable telephony resources. This non-exhaustive feature list would go a very long way to stopping the DoS attacks: QoS, per flow rate limiting (Micro-flow policing), aggregate rate limiting, IPsec VPNs, 802.1X network admission control (NAC), network intrusion detection (IDS) and prevention (IPS), firewalls (PIX and FWSM), access lists, IP and MAC spoofing prevention (IP Source Guard), DHCP spoofing and starvation prevention (DHCP Snooping), ARP poisoning and flooding prevention (Dynamic ARP Inspection), per port MAC restrictions (Port Security). While these Cisco network infrastructure features can be used to help protect any vendors IP Telephony system, a few of them provide additional value when using Cisco IP Telephony. For example, Cisco Discovery Protocol automates the configuration of VLANs on the Unified IP Phones, which not only reduces the complexity of configurable but also reduces your exposure to security vulnerabilities as well (802.1Q trunks can be a point of exposure if not properly administered/pruned...CDP eliminates the need for manual configuration of the 802.1Q properties). 802.1X port authentication also works best when using Cisco Unified IP Phones with Cisco Catalyst switches.

1.6.3 Confidentiality and Privacy (Packet Sniffing) Vendor Response Requirements Briefly described any embedded features/functions in the proposed IPTS that will preserve communications confidentiality and privacy. Indicate if control signaling and/or bearer communications signaling is encrypted at the call control, voice client, and media gateway elements to counter packet sniffing attempts.

Cisco Response: Cisco Unified IP Phones, Unified CallManager, gateways, Unity, IP Contact Center (SCCP only), and SRST gateways support bi-directional encryption of SCCP and SIP signaling through the use of TLS. In addition, bi-directional encryption of media is maintained from phones to other phones, gateways, IP Contact Center, Unity, and SRST gateways through the use of SRTP. In addition to encrypting signaling and media, per packet integrity and authentication is employed. Without per packet integrity and authentication, confidentiality of signaling and media is meaningless. All administrative web application traffic and LDAP directory requests are also encrypted using SSL. With Unified CallManager 4.0, Cisco introduced encryption of the RTP stream. Secure RTP, or sRTP, is an IETF standard: rfc3711. The entire packet contains an HMAC-SHA-1 authentication tag and the RTP payload is encrypted using AES-128-CBC.



Of mere trivial interest, an SRTP packet is virtually indistinguishable from an RTP packet. They have the exact same header information. SRTP contains the 4 byte authentication tag but a packet decoder wouldn’t be able to separate that from the end of the actual media. Other than playback, the only way to computationally determine if a packet is encrypted is to compute the statistical randomness of the packet. If it’s statistically random, it might be encrypted. That’s the best you can come up with. Media encryption can be enabled on a global or on a per phone basis. It is supported on the Unified IP Phone 7940G, 7941G, 7941G-GE, 7960G, 7961G, 7961G-GE, 7970G and 7971G-GE, on select IOS gateways running 12.3(11)T, and on Unity 4.0(4). Additional devices will support media encryption in the future. Whether or not a call is encrypted is part of the capabilities exchange during call setup. If both endpoints in a call are capable of media encryption, then the call will automatically be encrypted. End users are notified of this by the presence of a lock icon on the phone display. Configuration of TLS and SRTP is done through a single parameter. Setting a phone, gateway, or application to “Encrypted” mode enables both authenticated and encrypted signaling over TLS and authenticated and encrypted media over SRTP. In Unified CallManager 4.0, “Authenticated” was a viable mode indicating that the phone as able to sign and authenticate packets but not encrypt them. Beginning with Unified CallManager 4.1, all phone models capable of authentication are also capable of encryption and, thus, “Authenticated” mode is no longer an option. Currently, call encryption terminates at the egress of the network, whether that be over the PSTN or IP to another Unified CallManager Cluster, IP-IP Gateway, MTP, etc. A patent has been filed for maintaining an SRTP call over the PSTN using an ISDN “B” channel. Emerging protocols are needed for media encryption across organizational trust boundaries.

1.6.4 Physical Interfaces Vendor Response Requirements Are there separate physical network interfaces to IPTS administration, control, and voice transmission signaling functions?

Cisco Response: Each Unified CallManager server supports a single physical interface for administration, control, and voice signaling. However, there are a number of native features to Unified CallManager that limit security exposure as well as features to the hardware that provide a level of separation for administration traffic. In addition, the web application interfaces, signaling, console, and backend services support authentication, so although a single interface is shared, access to its resources won’t necessarily be permitted.



It’s also possible to enable the iLO (Integrated Lights Out) port—a separate physical port--on the MCS server’s to allow for a remote authenticated and encrypted access to the OS console. With this addition configuration, the Unified CallManager OS could be configured to use access lists to block all administration-related connections to the non-iLO port, thus separating administrative traffic from end-user, backend, and API traffic.

1.6.5 Root Access Vendor Response Requirements Is there direct Root access to the IPTS common control?

Cisco Response: CallManager 5.0 does not provide “root” access. Instead it has an IOS-like CLI that limits access to the platform’s configuration. Authenticated “root” access is permitted on the Unified CallManager 4 OS, but by default it is only available via physical access to the console. However, it is possible to separate management of event logs and administration of the OS, so a rogue “root” account couldn’t cover their tracks by removing logs. It also prevents the “root” user from viewing the security log.

2.0 IPTS Network Port Capacity Requirements The proposed IPTS must be capable of supporting port capacity requirements for the HQ facility and remote branches. It must also be capable of supporting future VoiceCon growth requirements at HQ and RO facilities. 2.1.0 Port Capacity Requirements The equipped port capacity of the proposed VoiceCon HQ IP Telephony System at time of installation and cutover must support of a mix of IP telephones, analog telephones, facsimile terminals, modems, local central office trunk circuits (analog and digital, long distance trunk circuits [digital, only], and private network trunk circuits [IP]). In support of general communications requirements, VoiceCon facilities will have a sufficient number of wiring closets distributed throughout each facility to satisfy ANSI/EAI/TIA 569 structured cabling specifications for voice and data communications. Wiring closets will be interconnected based on requirements of the selected system. The entrance facility (trunk connect panel), main telecom equipment room, and Main Distribution Frame (MDF) for each facility are located off the entrance lobby. It will be the responsibility of the contractor to provide all cross connects between labeled 110 terminal blocks in each wiring closet and the demarc or "smart jack" and their equipment. The following sections describe the port capacity requirements for each of the VoiceCon network locations. Satisfying these stated port capacity requirements is a MANDATORY requirement 2.1.1 HQ Facility



The HQ location is a four-floor facility that will support at time of system installation and cutover the following station equipment: * 1040 desktop IP station instruments; * 100 PC client softphones (including three attendant operator positions); * 10 IP audio conferencing units; * 26 analog telephones including 5 used for Power Failure Transfer Station operation; * 12 facsimile terminals; * 12 data modems. Station equipment is uniformly distributed within and across the four floors of the building. There are ten (10) wiring closets per floor, and one (1) main equipment room on the first floor. See Table 1 for station summary. 2.1.2 RO Facility The RO facility will be a two floor facility that will support: * 205 desktop IP station instruments; * 21 PC client softphones (including two (2) attendant operator positions); * 4 IP audio conferencing units; *10 analog telephones including 2 used for Power Failure Transfer Station operation; * 5 facsimile terminals; * 5 data modems. Station equipment is uniformly distributed within and across the two floors of the building. There are ten (10) wiring closets per floor, and one (1) main equipment room on the first floor. See Table 1 for station summary. 2.1.3 SB1 (Large) The SB1 facility will be a single floor facility that will support: * 75 desktop IP station instruments * 10 PC client softphones * 4 audioconferencing units; * 5 analog telephones including 2 Power Failure Transfer Stations; * 2 facsimile terminals; * 4 modems. All line station equipment will be equally distributed across the single floor of the building. There will be two (2) wiring closets, and one (1) main equipment room. See Table 1 for station summary. 2.1.4 SB2 (Medium) The SB1 facility will be a single floor facility that will support: * 37 desktop IP station instruments;



* 5 PC client softphones * 2 IP audioconferencing units; * 2 Analog station used as Power Failure Transfer Stations; * 2 facsimile terminals; * 2 modems. All line station equipment will be equally distributed across the single floor of the building. There will be two (2) wiring closets, and one (1) main equipment room. See Table 1 for station summary. 2.1.5 SB3 (Small) The SB3 facility will be a single floor facility that will support: * 8 Deskop IP station instruments; * 1 Analog station used as a Power Failure Transfer Station; * 1 facsimile terminal. All station equipment will be equally distributed across a single room on the main floor of the building. There will be one (1) wiring closet/equipment room. See Table 1 for station summary. Table 1: VoiceCon Equipped Station Requirements IP Station IP Station IP Att. IP Analog Analog Fax Modem

Instrument Softphone SoftconsoleAudio-Conf. Standard PFTS Terminal Device

HQ 1040 97 3 10 21 5 12 12 RO 205 19 2 4 8 2 5 5 SB1 75 9 1 4 3 2 2 4

SB2 37 5 0 2 0 2 2 2 SB3 8 0 0 0 0 1 1 0

2.2 Equipped Voice Terminal Requirements VoiceCon requires the following mix of wired and installed desktop IP telephone instruments (Table 4). Note: Descriptions of Desktop IP individual voice terminal types can be found in RFP Section 4

Table 4: VoiceCon Desktop IP Telephone Instruments Requirements



Facility Economy Administrative Professional Executive HQ 50 140 800 50 RO 15 30 150 10 SB1 8 10 55 2 SB1 3 8 25 1 SB3 0 1 7 0

Cisco Response: Cisco complies. The required equipped station quantities can be supported and have been quoted in this proposal.

2.3 Trunk Circuit Requirements The VoiceCon HQ and RO facilities will each have a combination of local, long distance, and private network trunk circuits. The SB facilities will each have a limited number of analog trunk circuits, but all long distance calls will be routed through the HQ facility. All facilities will also have PFTS circuits. All local digital trunks must be able to support a combination of inbound DID service and two-way CO trunk services. All long distance calls placed from a SB facility will be routed via the LAN/WAN through the HQ facility for PSTN trunk access. The following table summarizes HQ facility trunk circuit requirements for each of the four VoiceCon design configurations. Table 5: VoiceCon IPTS Network Equipped Trunk Port Requirements

Per Incremental Location

T-1 Digital Local Inbound/Outbound

T-1 Digital Long

Distance Analog (PFTS) 2-way GS/LS

HQ 6 7 5 25

RO 2 2 2 10

SB1 1 0 2 5

SB2 0 0 2 10

SB3 0 0 1 5



VoiceCon will engineer its WAN trunk circuits to support compressed voice traffic (G.729A algorithm voice codecs) among all IPTS network facilities inter-facility voice traffic. Any additional PSTN trunk circuits required to support local survivability requirements must be identified. Necessary common equipment must be included in the system configuration and pricing proposals and identified as such. Vendor Response Requirements Confirm that the proposed IPTS network solution satisfies the stated trunk circuit requirements; support of centralized long distance trunk resources at the VoiceCon HQ facility for the SB facilities; and automatic alternate routing of calls among all VoiceCon facilities across the WAN and PSTN.

Cisco Response: Cisco complies. The required equipped trunk ports can be supported and have been quoted in this proposal. The following table shows the actual equipped quantities proposed by configuration.

Location T1 (local Inbound/ Outbound)

T1 Long Distance

Analog (PFTS)

2-way GS/LS FXO

HQ 7 7 5 28 RO 2 2 2 10 SB1 1 1 2 8 SB2 0 0 2 10 SB3 0 0 2 6

Note: The GS/LS emergency trunks have been provisioned as FXO trunks on the gateway routers. Cisco Gateways Cisco currently offers a wide variety of different gateway products for connecting an IP telephony network to a PBX or the Public Switched Telephone Network (PSTN) or to legacy devices, such as analog phones, fax machines, modems, etc. This affords VoiceCon a tremendous amount of flexibility, depending on future capacity and application requirements. These gateways can have analog DS-0 interfaces using FXS, FXO, or E&M signaling or digital BRI/PRI, T1/E1 protocols. In addition, these gateways support standard features such as ANI, DID, DOD, etc. Cisco offers three basic gateway options:

1. Gateway modules for the Catalyst switch family such as the 48-port FXS, or the 6-port T1/E1 and 24-port FXS port adapters for the Catalyst 6500 Communications Media Module.

2. Gateway modules for Cisco routers, which include FXS, FXO, T1, E1, BRI, PRI, TIE, and CAMA.



3. Standalone voice gateways (single purpose devices), the VG224 and VG248 for FXS, the ATA 186 and ATA 188 for FXS, and the Digital Port Adapter for connecting to legacy voice mail systems.

Voice Gateways Used In This Proposal: Based on the requirements of this RFP, several gateways have been included in the proposal:

1) Cisco Integrated Services Routers (ISRs), 2821 and 2851 series with VWIC, or WIC or high density modules for T1, FXO or FXS ports.

2) VG224 or VG248 Analog Phone Gateways to provide 24 or 48 ports of analog FXS

The following is a brief description of the capabilities of each of the gateway components included in this proposal:

a. Cisco Integrated Services Routers (2800 and 3800 Series):

Customers of all sizes want opportunities to converge best-of-class data, voice, and security services into a single system that enables rapid services deployment, as well as opportunities to protect, grow, and optimize their businesses. With Cisco® Integrated Services Router platforms, enterprise branch, commercial offices and small or medium-sized offices can use the industry's broadest, most comprehensive voice and security services, directly embedded and integrated inside the industry's leading routing platform for maximum performance and resiliency. Cisco Integrated Services Router platforms (Cisco 2801, 2811, 2821, 2851, 3825, and 3845) provide the appropriate-sized solution for the smallest to largest customers, scaling to meet the most demanding enterprise environments while providing the performance and architecture for services available today and in the future. Cisco ISR’s have been included in this proposal to provide PSTN gateways at headquarters, and for PSTN gateways and SRST at the remote locations. Cisco Integrated Services Router platforms embed voice and security functions directly inside the router, enabling customers to deploy advanced services simply by installing digital signal processors (DSPs) and advanced integration modules (AIMs) for IP telephony conferencing, voice gateways, Cisco Unity™ Express voice mail and automated attendant, as well as industry-standard security. For call processing, customers can enable the



company's award-winning Cisco Unified CallManager Express solution as part of Cisco IOS® Software, and easily reconfigure the same software at any time to support Cisco Survivable Remote Site Telephony (SRST) for centralized call processing with Cisco Unified CallManager, the industry's leading solution for enterprise-class IP telephony. By embedding proven voice technologies inside the platform, Cisco Integrated Services Router platforms free up integrated modular slots, enabling customers to take advantage of new high-speed slots such as the high speed WAN interface card (HWIC) and enhanced network module (NME) for additional services, interfaces, and densities. The innovative extension voice module (EVM) slot on both the Cisco 2821 and 2851 routers provides significant increases in analog and Basic Rate Interface (BRI) voice densities within a single platform, scaling to as many as 24 foreign exchange station (FXS) interfaces and 12 foreign exchange office (FXO) interfaces (these EVM modules also are supported on the Cisco 3825 and 3845 routers). All Cisco Integrated Services Router platforms natively support voice interface cards (VICs) in all HWIC slots and use the optional packet voice DSP modules (PVDMs), which are installable on the motherboard of the router itself. The following two figures show IP Communications services available on the Cisco 2851 and Cisco 3845, respectively.

IP Communications Services Available on a Cisco 2851 Integrated Services Router

IP Communications Services Available on a Cisco 3845 Integrated Services Router

FEATURES TABLE Cisco IP Communications features on Cisco Integrated Services Router platforms (Cisco 2801, 2811, 2821, 2851, 3825, and 3845) include advanced features and services for implementing networked, loosely-



coupled, and semiautonomous business solutions. Some of these features include call processing, call control protocols, quality of service (QoS), analog and digital interfaces, queuing, scripting, voice mail, and automated attendant. The table below shows some of the key IP Communications features found on the Cisco Integrated Services Router platforms.

Key IP Communications Features on Cisco Integrated Services Router Platforms

Telephony Advanced Services

QoS, Queuing, Call Control

Cisco IP Communications

Analog FXS, FXO, E&M, Centralized Automated Message Accounting (CAMA) and direct inward dialing (DID) telephony interfaces

Motherboard-based DSP conferencing (up to 50 sessions)

Low Latency Queuing (LLQ), Class-Based Weighted Fair Queuing (WFQ), CB-WFQ, Class-Based CB-Weighted Random Early Detection

Cisco Unified CallManager Express (24, 36, 48, 96, 168, and 240 phone support)

Digital T1, E1, and BRI telephony interfaces

Motherboard-based DSP transcoding (up to 128 sessions)

Media Gateway Control Protocol (MGCP), H.323, and Session Initiated Protocol (SIP) call control protocols

Cisco Unity Express (8 hour, 4 port AIM or 100 hour, 8 port network module)

Integrated analog to digital channel bank

Integrated access (data WAN and voice termination)

Advanced and local voice busy out (AVBO and LVBO)

Cisco Unified CallManager gateway

Module line-powered Ethernet switch

Voice extensible markup language, Tool Command Language scripting support

Resource Reservation Protocol, Call Admission Control (CAC)

Cisco Automated Attendant on Cisco Unity Express

AVBO AND LVBO Built-in support for future video capabilities

Class of service (COS) to type of service (TOS) mapping

Cisco SRST (24, 36, 48, 72, 96, 144, 480, 336, and 720 phone support)

APPLICATIONS Cisco Integrated Services Router platforms are the foundation for enterprise branch offices and commercial offices deploying IP telephony with SRST for failover call processing. In this deployment model, voice mail and automated attendant services can be delivered directly inside the Cisco integrated services router using Cisco Unity Express or delivered centrally using Cisco Unity software. Customers can also implement Secure SRST to enable authentication and encryption support for both signaling and media transmission during a WAN outage. Loosely coupled offices or semiautonomous offices are ideal locations for the "right size" IP telephony services within a Cisco integrated services router, which delivers the security and routing requirements for that same location.



Integrating Cisco Unified CallManager Express and Cisco Unity Express within the Cisco integrated services router management simplifies service and maintenance. The figure below shows the Cisco Integrated Services Router platforms used for IP telephony in branch, commercial or small offices. Cisco Unified CallManager Express can also be transparently integrated with a central Cisco Unified CallManager for maximum configuration flexibility.


PVDM Slots on Motherboard Embedded modularity using any Cisco PVDM2 product (PVDM2-8, PVDM2-16, PVDM2-24, PVDM2-32, or PVDM2-64) on any Cisco integrated services router platform motherboard delivers conferencing, transcoding, and voice termination without the need to use a network module or AIM. Installing Cisco PVDM2 products inside the Cisco integrated services router provides these services for both voice over IP traffic along with time-division multiplexing (TDM) traffic. TDM interfaces can be terminated by using VICs, VWICs or an EVM (EVM is supported only on the Cisco 2821, 2851, 3825, and 3845 routers) installed natively on the Cisco integrated services router.

Dedicated EVM Slot on Cisco 2821 and Cisco 2851 Routers Exceptional analog and BRI telephony densities are achieved through the use of a specialized EVM, which frees expansion network module slots for additional services and increased densities of voice, LAN and WAN interfaces. Using PVDM slots on the motherboard in conjunction with the EVM card, customers have the option to offer voice services within the same platform that one would normally deploy for security and routing without the need to add a second router.

Media Authentication and Encryption



Media authentication and encryption features on Cisco's portfolio of Access Routers ensures that voice conversations terminating on either TDM or analog voice gateway ports are protected from eavesdropping. These reliable, scalable features provide a secure environment for IP communications over a LAN (local area network) or WAN (wide area network).

Dedicated Echo Cancellation Slot on 2nd Generation Multiflex Trunk Voice/WAN Interface Cards The Cisco 2nd Generation 1 and 2 Port T1/E1 Multiflex Trunk Voice/WAN Interface Cards (VWIC2-MFT family) have an on-board slot for a Multiflex Trunk Dedicated Echo Cancellation Module (EC-MFT-32 and EC-MFT-64). These optional daughter cards provide a dedicated hardware resource that runs Cisco's Enhanced ITU-T G.168 Echo Cancellation feature, which can be used in select network conditions when the existing voice resources are constrained. The Dedicated Echo Cancellation module's processing and memory resources enables the echo canceller to be configured with an extended 128ms echo tail buffer, providing a robust echo cancellation performance for demanding network environments.

Cisco Router Network Modules and Port Adapters: The Cisco® IP Communications voice/fax network modules provide enterprises, managed service providers, and service providers the ability to directly connect the public switched telephone network (PSTN) and traditional telephony equipment (private branch exchange [PBX], key system, analog telephones, fax machines, etc.) to existing Cisco voice gateway routers. This set of Cisco IP Communications voice/fax network modules delivers the most versatile combination of analog and digital voice and data capabilities in a single network module. The Cisco IP Communications voice/fax network modules support either one or two Cisco voice interface cards (VICs) or Cisco voice/WAN interface cards (VWICs) and install into network module slots for the Cisco voice gateway routers. The Cisco VICs are daughter cards that install into the network modules and provide the interface to the PSTN and to telephony equipment (PBX, key systems, fax machines, phones). The Cisco VWICs are daughter cards that provide the interface to the PBX, PSTN, and/or WAN.

2-port FXO and 4-port FXS 2-port T1 Adapter (on right side)

VICs include 2-port foreign exchange station (FXS), direct inward dial (DID),



foreign exchange office (FXO), and E&M analog interface cards. Also available are 4-port FXS and 4-port FXO cards and a 2-port ISDN Basic Rate Interface (BRI) digital interface card providing --40V phantom power. These cards cover the entire range of analog connectivity options along with user-side and network-side digital BRI connections. Cisco VWICs include 1- and 2-port T1 and E1 interface cards with optional drop-and-insert capability along with a G.703 interface card option. These cards cover a full range of digital voice and WAN connectivity options and provide connectivity to the world's PBXs, PSTNs, and Post, Telephone, and Telegraph (PTT) organizations.

Cisco High-Density Analog and Digital Extension Module for Voice and Fax

The Cisco High-Density Analog and Digital Extension Module for Voice and Fax (EVM-HD) provides enterprises, managed service providers and service providers the ability to directly connect public-switched telephone networks (PSTNs) and existing telephony equipment to Cisco 2821, Cisco 2851, Cisco 3825, and Cisco 3845 integrated services routers. With support for up to 24 total voice and fax sessions, the Cisco High-Density Extension Module is ideal for networks with high analog (foreign exchange station [FXS], direct inward dialing [DID], and foreign exchange office [FXO]) and digital (Basic Rate Interface [BRI]) call-capacity requirements.

The Cisco High-Density Analog and Digital Extension Module for Voice and Fax (EVM-HD) is an industry-leading voice and fax interface module for Cisco 2821, Cisco 2851, Cisco 3825, and Cisco 3845 integrated services routers. It helps enable packet voice technologies with support for voice over IP (VoIP), including H.323, Media Gateway Control Protocol (MGCP) and Session Initiation Protocol (SIP); voice over Frame Relay (VoFR), and voice over ATM (VoATM), including ATM Adaptation Layer 2 (AAL2) and AAL5. The Cisco High-Density Extension Module allows Cisco 2821, Cisco 2851, Cisco 3825, and Cisco 3845 routers to connect directly to the PSTN and existing telephony equipment (for example, private branch exchange (PBX), Key system, analog telephones, and analog fax machines) through standard analog (FXS, DID, and FXO) and digital (BRI) interfaces. The High-Density Extension Module supports telephony toll bypass, new packet telephony applications, and full gateway integration within the Cisco AVVID (Architecture f or Voice, Video and Integrated Data) IP telephony design. With support for up to 24 total



voice and fax sessions, the Cisco High-Density Extension Module is ideal for networks with high analog and BRI call-capacity requirements.

Packet voice DSP modules (PVDM2s) are used in combination with the Cisco High-Density Extension Module baseboard and its expansion modules. PVDM2s support multiple voice codecs, fax, conferencing, and transcoding services. PVDM2s are purchased separately and installed in the digital signal processor (DSP) module slots located inside the Cisco 2821, Cisco 2851, Cisco 3825, and Cisco 3845 integrated services routers.

2) Cisco VG 224 and VG 248 Analog Phone Gateways:

VG 224 Analog Phone Gateway

VG248 Analog Phone Gateway

The Cisco VG 224 and VG 248 Analog Phone Gateways are high-density 24- and 48-port gateways for analog phones, fax machines, modems, and speakerphones within an enterprise voice system based on Cisco Unified CallManager. These gateways support analog phone lines to be used as extensions to the Cisco Unified CallManager or Cisco Call Manager Express system in very compact 19-inch rack-mount chassis. The Cisco VG224 and VG248 provide the following features:

• Caller ID—The Cisco VG248 supports caller ID (both name and number), so users can tell who is calling before answering the phone.

• Message-waiting light—The Cisco VG248 supports two methods of analog light activation: high-DC voltage (message-waiting indicator light) and frequency-shift-key (FSK) messaging, as well as stuttered dial tone for phones without a visual indicator. These schemes are used by private branch exchange (PBX) systems and central offices, respectively.

• Call waiting—When on a call, if a new call comes in the user hears an audible tone and can "click over" to the new caller.

• Caller ID on call waiting—The user can see who is calling before deciding whether to take the new call.



• Transfer—Both blind and supervised transfers are supported, using the standard Bellcore flash hook method.

• Conference—Conference calls can be initiated from an analog phone using flash hook or feature access codes, and the Cisco Unified CallManager directs the devices to a media resource for the conference call. Up to six parties can participate.

• Feature access codes—More advanced features can be activated using feature access codes.

• Speed dial—A user can set up commonly dialed numbers using the Cisco Unified CallManager Web interface and then dial these numbers directly from an analog phone.

• Call forward all—Calls can be forwarded to a number within the dial plan.

• Redial—A simple last-number redial can be activated from analog phones connected to the Cisco VG248.

• Cisco IP SoftPhone support—SoftPhone functionality is supported with analog phones.

Fax and Modem Connectivity The Cisco VG248 supports legacy fax machines and modems. When using fax machines, the Cisco VG248 uses the Cisco fax-relay technology to transfer faxes across the network with high reliability using less bandwidth than a voice call. Any modems can be connected to the Cisco VG248.

2.3.1 ISDN PRI Services All installed VoiceCon T-1 trunk circuits must support ISDN PRI features and functions for both local and long distance exchange carrier transmission services. Vendor Response Requirements Confirm that the proposed T-1 trunk circuit interfaces support ISDN PRI capabilities. 2.4 IPTS Network Growth Requirements VoiceCon anticipates that station capacity requirements at the HQ and RO facilities will increase approximately 50% for the expected installed life of the proposed IPTS network solution. Port capacity growth requirements at the SB1 and SB2 faciilties are anticipated to increase by about 20%; no growth is anticipated at the SB3 facility. Vendor Response Requirement: Confirm that the proposed IPTS solution can satisfy VoiceCon station port growth requirements and associated trunk growth requirements at its HQ, RO, SB1 and SB2 facilities without replacing any hardware equipment at time of initial system installation and cutover. Hardware additions are permissible to support incremental port interface requirements.



Cisco Response: Cisco complies. The proposed Cisco IP telephony solution easily supports the initial deployment of 1,610 users, and can scale to 2,500 users (50% growth) without any change or additions to the solution as proposed. Additional gateway ports would probably be needed, but can easily be added to the core configuration. Future growth to 30,000 users is possible by adding more of the same components without any changes to the underlying architecture or replacement of cabinets as is common with legacy systems.

3.0 Port Interface and Traffic Handling Requirements The proposed IPTS network solution must be able to support a mix of TDM/PCM and IP ports. For traffic design engineering calculations assume the following traffic requirements:

1. The average busy hour traffic for IP desktop station users will be rated at 10 CCS @ P.01. Assume a traffic mix pattern of 30% intra-network calls, 15% outgoing local trunk calls, 25% outgoing long distance trunk calls, and 30% incoming DID trunk calls.

2. Analog telephone station busy hour traffic will be rated at 3 CCS @ P.01. Assume a traffic mix pattern of 70% inter-network calls and 30% outgoing local trunk calls. All analog telephone station calls will be subject to toll restrictions.

3. Assume that busy hour traffic is rated at 36 CCS @P.01 for each of the following port types: all PSTN and WAN trunk circuits; attendant consoles; modems; audioconferencing units; facsimile terminals; voice mail ports.

Vendor Response Requirement: The proposed system must design and engineer their system to support the above traffic assumptions. Confirm you have satisfied this requirement.

Cisco Response: The proposed system has been engineered to meet these requirements. The IP LAN infrastructure that forms the switching matrix in a Cisco IP communications system is completely non-blocking.

Cisco IP Communications systems are built on a 10/100 Mb switched Ethernet LAN infrastructure. All end points connect to a 10/100 Mb port and have full access to the bandwidth. Cisco IP Communications systems are a distributed LAN-based IP system that uses packet switching. It is not TDM-based so there is no actual limit to the number of simultaneous conversations supported. The practical limit is 7500 Unified IP Phones per MCS 7845 Unified CallManager server in a non-blocking configuration, and up to 30,000 Unified IP Phones in a Unified CallManager cluster.

3.1 Circuit Switched Network Design



The proposed IPTS solution must support a variety of peripheral ports and switched connections. Although it is not required to support traditional digital voice terminal equipment, the IPTS must support analog communications devices. Switched connections involving non-IP ports may be handled using a circuit switched network, media gateways/Ethernet switches, or a combination of both methods. Vendor Response Requirement: If the proposed IPTS network solution includes integrated circuit-switched hardware equipment, then briefly describe the characteristics of the offering. Include, at minimum, the following information: hardware cabinet description; CCS @P.01 rating; center stage switch and local TDM bus time slot/talk slot capacities; interswitch link capacities; all redundant design elements and level of redundancy.

Cisco Response: Not Applicable. 3.2 Peer-to-Peer Communications for IP Station to IP Station Calls All two-party voice calls between IP desktop stations located at VoiceCon facilities must be handled exclusively over the LAN/WAN infrastructure without any circuit switched connections. This is a Mandatory requirement. Vendor Response Requirement: Confirm that your proposed system satisfies this Mandatory requirement.

Cisco Response: Cisco complies. Cisco IP Communications systems are end-to-end IP communications systems. All Unified IP Phone to Unified IP Phone voice calls are IP-based on the LAN/WAN infrastructure with no TDM components involved (with the obvious exception of calls that are alternately routed to the PSTN during WAN congestion or failure conditions).

3.2.1 IP Station Discovery How do IP communications devices learn about their voice VLAN, including IP addresses, default gateways, call controller, TFTP server, QoS settings, VLANs, and other parameters. Does the proposed system solution employ proprietary protocols for IP communications devices to learn their voice VLAN or is an industry standard, such as Dynamic Host Control Protocol (DHCP) used?

Cisco Response: The Cisco IPT phone use standard DHCP to obtain IP address, default gateways, TFTP server and other IP network related parameters. When connected to CDP enabled switches, the phone are optionally able to get Voice VLAN, QOS settings and extended PC Class of Service settings. If the IP Phones are not connected to a CDP enabled switch, the phones do allow for hard coding of the voice VLAN and to ensure proper VLAN



association for voice data. Call controller information is obtained from the TFTP server specified in the DHCP lease. The administrator can optionally hard code the TFTP server on a phone by phone basis.

3.2.2 IP Station Power over Ethernet (PoE) VoiceCon requires that the power option to support IP telephones conform to IEEE 802.3af Power over Ethernet (PoE) standards. Vendor Response Requirement: Confirm that the proposed IPTS solution supports the IEEE 802.3af specification for in-line of IP telephone equipment. Describe current, future and retrospective compatibility of all proposed equipment. If 802.3af is not supported, identify the PoE implementation being proposed.

Cisco Response: The Cisco IPT Stations specified in this RFP support IEEE 802.3af in-line power with the exception of the 7936 conference station. The 7936 requires a power insertion module for power. Additionally, the 7906, 7961 and 7970 phones support the Cisco Inline power standard. This will allow investment protection for customer with existing switches that support Cisco’s pre-802.3af inline power generation.

3.2.3 IP Station QoS Vendor Response Requirement: Describe the proposed IPTS solution’s capabilities to provide Layer 2 and Layer 3 QoS to IP stations to ensuring end-to-end quality of service. Include in the response what industry standards are deployed.

Cisco Response: The Cisco IP Stations perform layer 2 and layer 3 packet marking. At layer 2, the IP Station will correctly set the COS value defined by the switch administrator. The IP station will inspect the data packets from any attached device and modify the COS value to the administratively defined value on the CDP enabled switch. The phones will also set the layer 3 DSCP value with industry standard EF for voice RTP traffic and AF31 (CS3) for signaling traffic. The IP station has the ability to rewrite the Layer 3 DSCP/TOS value in packets received on the data port to an administratively defined value automatically.

3.3 Multi-Party Conference Calls The proposed system must be able to support six party add-on conference calls among IPTS stations and off-network stations. The system must also support a minimum of three (3) off-network stations per multi-party conference call when required. The HQ IPTS must support a minimum of 20



simultaneous multi-party add-on conference calls (up to six parties per conference) and the RO IPTS a minimum of 10 simultaneous multi-party add-on conference calls (up to six parties per conference) Vendor Response Requirement: Briefly explain how multi-party add-on conference calls are handled if: 1) All parties are on-network IP stations; 2) There is a mix of on-network IP and off-network stations. The explanation should identify any and all hardware and software requirements necessary to support multi-party add-on conference call requirements. Specify if peripheral hardware equipment, e.g., conference bridge servers, is required.

Cisco Response: Cisco Unified CallManager supports both Meet-Me conferences (up to 100 participants) and Ad Hoc conferences (up to 6 participants). Meet-Me conferences allow users to dial in to a conference. Ad Hoc conferences allow the conference controller to let only certain participants into the conference. In the proposed configuration, conference resources are located at the Unified CallManager cluster at Headquarters as well as the DSP modules in the ISR’s in the Regional Office and branch offices.

Cisco Unified CallManager supports multiple conference devices to distribute the load of mixing audio between the conference devices. A component of Cisco Unified CallManager called Media Resource Manager (MRM) locates and assigns resources throughout a cluster. The MRM resides on every Cisco Unified CallManager and communicates with MRMs on other Cisco Unified CallManager servers.

Both hardware and software conference bridges can be active at the same time. Software and hardware conference devices differ in the number of streams and the types of codec that they support. For software conference devices, you can adjust the number of streams. Hardware conference devices, however, support a fixed number of streams.

Hardware-enabled conferencing provides the ability to support voice conferences in hardware. Digital Signaling Processors (DSPs) located on devices such as Catalyst Voice Modules convert multiple Voice over IP Media Streams into TDM streams that are mixed into a single conference call stream. The DSPs support both Meet-Me and Ad Hoc conferences by the Cisco Unified CallManager.

Software conference devices support a variable number of audio streams. You can create and configure a software conference device within a Unified CallManager server (or a dedicated conferencing server) and select the number of full-duplex audio streams that the device supports. To calculate the total number of conferences that a device supports, divide the number of audio streams by three. The maximum number of audio streams is 128.



Meet-Me conferences require that a range of directory numbers be allocated for exclusive use of the conference. When a Meet-Me conference is set up, the conference controller selects a directory number and advertises it to members of the group. The users call the directory number to join the conference. Anyone who calls the directory number while the conference is active joins the conference.

The conference controller controls Ad Hoc conferences. When you initiate an Ad Hoc conference, Cisco Unified CallManager considers you the conference controller. In an Ad Hoc conference, only a conference controller can add participants to a conference. The conference controller can add up to the maximum number of participants specified for Ad Hoc conferences to the conference provided that sufficient streams are available on the conference device.

3.4 VoIP Overflow Traffic If available WAN circuits connecting the HQ, RO and all SB facilities are busy, call admission control levels are reached, or QoS levels are not satisfied on-network voice traffic must be able to automatically overflow to PSTN trunk circuits. Vendor Response Requirement: Confirm that your proposed communications system supports overflow of voice traffic across VoiceCon locations if WAN links are not available or conditions are not acceptable. Also indicate if overflow traffic can revert back to the WAN if conditions permit.

Cisco Response: Cisco complies. The standard Automated Alternate Routing (AAR) feature provides a mechanism to reroute calls through the PSTN or other network by using an alternate number when Cisco Unified CallManager blocks a call due to insufficient WAN bandwidth (or loss of WAN connectivity). With automated alternate routing, the caller does not need to hang up and redial the called party. The AAR group represents the dialing area where the line/directory number (DN), the Cisco voice mail port, and the gateway are located.

3.5.0 Port Interface Circuit Cards For each of the following port types, provide a brief description of the proposed port interface circuit card(s) and/or media gateway equipment included with the proposed IPTS to support analog, digital, and IP ports. Include in the descriptions below the number of port interface terminations for each port circuit card, and the number of available gateway channels for each media gateway unit. 3.5.1 IP Telephones (desktop instrument and PC client softphones, including Attendant Console Position) & IP Audioconferencing Units



Vendor Response Requirement: Provide a brief description how all IP telephone types are logically and physically supported by the common control call telephony server. If direct call control signaling via the Ethernet LAN/WAN is not supported identify all intermediary carrier, signaling interface and/or media gateway equipment that is required.

Cisco Response: Cisco Unified IP Phones are logically supported by Cisco Unified CallManager which performs the gatekeeper functions for Unified IP Phone registration, as well as call control, feature support, etc. Physically, Cisco Unified IP Phones attach to Catalyst® (or other industry standard) 10/100 or 10/100/1000 Ethernet switch ports with a simple RJ-45 connection, and configure themselves to the IP network via the Dynamic Host Control Protocol (DHCP) and TFTP.

3.5.2 Analog telephones Vendor Response Requirement: Provide a brief description how analog telephones are logically and physically supported by the common control call telephony server, identifying all intermediary hardware elements necessary for control signaling transmission. Specify the number of circuit terminations per circuit board/module/media gateway.

Cisco Response: Analog phones can be supported on standard FXS ports provided by a variety of Cisco gateways. The following Cisco products provide FXS support:

• Cisco Analog Telephone Adapter (ATA) 186 and 188 (2 ports) • Cisco Voice Gateway (VG) 224 (24 ports) and VG248 (48 ports) • Cisco Voice Modules for Cisco routers (2, 4 or 8 ports) • Cisco 24-Port FXS Port Adapter for the Cisco Communications Media

Module (Catalyst 6500 Series) • Cisco Catalyst 6000 Family Analog FXS Interface Module (48 ports) • Cisco Catalyst 4000 Family Analog Gateway Module (8 ports)

3.5.3 Facsimile terminal Vendor Response Requirement: Provide a brief description how facsimile terminals are logically and physically supported by the common control call telephony server, identifying all intermediary hardware elements necessary for control signaling transmission. Specify the number of circuit terminations per circuit board/module/media gateway.

Cisco Response: Analog Fax support is provided by the gateways listed in 3.2.2 above.

3.5.4. Modem Vendor Response Requirement: Provide a brief description how modem terminals are logically and physically supported by the common control call telephony server, identifying all



intermediary hardware elements necessary for control signaling transmission. Specify the number of circuit terminations per circuit board/module/media gateway.

Cisco Response: Modems are supported by the gateways listed in 3.2.2 above.

3.5.5 Power Failure Transfer Station (PFTS) Vendor Response Requirement: Provide a brief description how analog telephone instrument Power Failure Transfer Stations (PFTSs) are logically and physically supported by the common control call telephony server, identifying all intermediary hardware elements necessary for control signaling transmission. Specify the number of circuit terminations per circuit board/module/media gateway.

Cisco Response: In the Headquarters and remote office locations, the Cisco High Density FXO extension modules included in this proposal support power failover. All FXO ports on these modules support power failure bypass. Power failure transfer stations are provided by Gordon Kapes power failure bypass units (8 ports each) in the remote offices.

3.5.6 GS/LS CO Trunk Vendor Response Requirement: Provide a brief description how GS/LS CO trunk circuits are logically and physically supported by the common control call telephony server, identifying all intermediary hardware elements necessary for control signaling transmission. Specify the number of circuit terminations per circuit board/module/media gateway.

Cisco Response: GS/LS trunk circuits are supported on standard FXO ports provided by Cisco gateways. The following products provide FXO support:

• Cisco FXO Voice Modules for Cisco routers (1 or 2 ports) • Cisco High Density Expansion Modules for Cisco Routers (3 or 8 ports)

3.2.7 DS1/T-1 Carrier Interface Trunk Vendor Response Requirement: Provide brief description how DS1-based T-1 carrier trunk circuits are logically and physically supported by the common control call telephony server, identifying all intermediary hardware elements necessary for control signaling transmission. Specify the number of circuit terminations per circuit board/module/media gateway.

Cisco Response: DS1 T-1 carrier trunk circuits are supported on standard T1 ports provided by a variety of Cisco gateways. The following products provide T-1 support:

• Cisco T1 Voice Modules for Cisco routers (1 or 2 ports) • Cisco 6-Port T-1 Port Adapter for the Cisco Communications Media

Module (Catalyst 6500 Series)



• Cisco Catalyst 6000 Family Voice Services Module (8 ports) • Cisco Catalyst 4000 Family T-1 Gateway Module (8 ports)

3.2.8 Other Trunk Interfaces VoiceCon may need at some future time additional analog trunk interfaces, specifically Auxiliary, FX, and E&M Tie Line. Vendor Response Requirement: Provide a brief description of how additional analog trunk interface requirements can be logically and physically supported by the common control call telephony server, identifying all intermediary hardware elements necessary for control signaling transmission. Specify the number of circuit terminations per circuit board/module/media gateway.

Cisco Response: Adding additional trunk interfaces to the configuration is simply a matter of adding modules to existing Cisco gateways that have open slots, or adding new Cisco gateways. Cisco gateway specifications, capacities, module density, etc., is covered in detail in Section 2.3 above, “Cisco Gateways”.



4.0.0. Voice Terminal Instruments The proposed communications system must be able to support a mix of analog and IP communications devices. VoiceCon will provide its own analog telephone instruments, fax terminals, and modems. 4.1 Regulation Requirements All single- and multi-line Unified IP Phones will be manufactured in accordance with Federal Communication Commission hearing aid compatibility technical standards contained in Section 68.316. and the Telecommunication Act of 1996. Vendor Response Requirement: Confirm the proposed telephone equipment satisfies these requirements

Cisco Response: Cisco complies. 4.2 Desktop IP Telephone Instruments VoiceCon has a requirement for several types of desktop IP telephone instruments:

• Economy • Administrative • Professional • Executive

Cisco Response: Cisco has proposed a mix of Unified IP Phones to meet the RFP requirements. These are native Unified IP Phones which support Cisco inline power and/or 802.3af (IEEE) inline power. The following specific Cisco Unified IP Phones were included in the response:

Economy Desktop IP Phone: Cisco Unified IP Phone 7906G Administrative IP Phone: Cisco Unified IP Phone 7931G Professional IP Phone: Cisco Unified IP Phone 7961G Executive IP Phone: Cisco Unified IP Phone 7970G Audio Conference Station: Cisco 7936 IP Conference Station PC Client Softphone Cisco IP Communicator Attendant Console Cisco Attendant Console

4.2.1 Economy Desktop IP Telephone Instrument The Entry model will be used in common areas. It should have, at minimum, the following design attributes and features/functions:

• 12 key dial pad • Single line appearance



• Hold button • G.711/G.729 voice codecs • Auto Self Discovery/DHCP • Echo Canceller • IEEE 802.af POE support

Vendor Response Requirement: Confirm that your proposed Economy model satisfies all of the stated requirements and provide a brief product description that includes an illustration/ photograph (PPT format, only) of the instrument. Indicate in your response any and all requirements not satisfied.

Cisco Unified IP Phone 7906G

Cisco Response: The Cisco Unified IP Phone 7906G has been quoted to meet this requirement. The Cisco® Unified IP Phone 7906G fills the communication needs of cubicle, retail, classroom, or manufacturing workers or anyone who conducts low to moderate telephone traffic. Four dynamic soft keys guide users through core business features and functions, while a pixel-based display combines intuitive features, calling information, and eXtensible Markup Language (XML) services into a rich user experience. The Cisco Unified IP Phone 7906G offers numerous important security features plus the choice of IEEE 802.3af Power over Ethernet (PoE), Cisco inline power, or local power through an optional power adaptor

Cisco Unified IP Phone 7906G Features

The Cisco Unified IP Phone 7906G is designed to grow with your organization. A dynamic, soft key-activated feature set helps enable the phone to keep pace with your requirements through regular software upgrades. Moves, adds, and changes are easy; users can simply pick up their phones and move to a new location anywhere on the network. The Cisco Unified IP Phone 7906G also provides accessibility features to those with special needs.

• Lighted Hold Key—Lights when pressed to put a call on hold and stays lit until the held call has been resumed, or flashes if one call is held while another is engaged; is dark when no calls are on hold.

• Lighted Menu Key—Lights when pressed to access voicemail messages, call logs, network settings, user preferences, corporate directories, and XML services; stays lit while menu items are active.



• Lighted Message Waiting Indicator—Lights when there is new voicemail and is visible on both the phone chassis and the handset; stays lit until new voicemail has been processed by the user.

• Graphical Display—Graphical monochrome display with resolution of 192 x 64 pixels provides a scrollable 3-line intuitive access to calling features and text-based XML applications; the Cisco Unified IP Phone 7906G also supports audio-based XML applications.

• Four Soft Key Buttons and a Scroll Toggle—Dynamically present calling options to the user; the scroll toggle bar allows easy movement through the displayed information.

• Network Features—Offers Cisco Discovery Protocol; IEEE 802.1 p/q tagging and switching.

• Volume Control—A volume-control toggle provides easy decibel-level adjustments of the handset, speaker, and ringer.

• Single-Position Foot Stand—Provides optimum display viewing and comfortable use of buttons and keys; the foot stand can be removed for wall mounting with mounting holes located on the base of the phone.

• Multiple Ring Tones—Offers more than 24 user-adjustable ring tones.

• American Disabilities Act (ADA) Features—Hearing-aid-compatible (HAC) handset meets the requirements set by the ADA; it also meets ADA HAC requirements for a magnetic coupling to approved hearing aids; the phone dialing pad also complies with the ADA.

• Signaling Protocol Support—Supported in Cisco Unified CallManager Versions 3.3(5)SR2, 4.1(3)SR3a, 4.2(1)SR1, and higher using Skinny Client Control Protocol (SCCP); supports both SCCP and Session Initiation Protocol (SIP) with Cisco Unified CallManager Version 5.0(2).

• Codec Support—Provides G.711a, G.711µ, G.729a, and G.729ab audio-compression codecs.

• Configuration Options—Provides provisioning of network parameters through Dynamic Host Configuration Protocol (DHCP).

• Voice Quality—Offers comfort-noise generation and voice-activity-detection (VAD) programming on a system basis.

4.2.2 Administrative Desktop IP Telephone Instrument The Administrative model will be used by station users who have executive management group call answering and coverage responsibilities. It should have, at minimum the following design attributes and features/functions:

• 12 key dial pad • Sixteen (16) programmable line/feature keys with soft label/status

indicators • G711, G729 and wideband, e.g., G.722, voice codecs • Auto Self Discovery/DHCP • Echo Canceller • QoS Support (802.1p/Q, DiffServ)



• Hold key • Last Number Redial key • Release key • Message Waiting/Call Ringing indicator(s) • Full Duplex Speakerphone • Speaker/Mute key • Volume Control keys/slide • High resolution, backlit, monochrome grayscale pixel-based, graphical

display screen with four (4) associated context sensitive soft feature labels (key, cursor, or navigator control)

• LDAP access • Stored Call Data (Last 10 numbers dialed/Last 10 incoming call

numbers) • Integrated Ethernet switch and two (2) RJ-45 connector interface ports

for 10/100 Mbps connectivity • Headset interface • IEEE 802.af POE support

The Administrative model must also be capable of supporting optional add-on key modules if an additional 12 programmable line/feature keys with soft label/indicator status is required at some future time. Vendor Response Requirement: Confirm that your proposed Administrative model satisfies all of the stated requirements. Provide a brief product description that includes an illustration/photograph (PPT format, only) of the instrument. Indicate in your response any and all requirements not satisfied. State which required feature-specific keys are not available, but softkey feature access can be used as an alternative.


Cisco Response: The Cisco Unified IP Phone 7931G has been quoted to meet this requirement. The Cisco Unified IP Phone 7931G meets the communication needs of retail, commercial, manufacturing workers, and anyone with moderate telephone traffic but also specific call requirements. Dedicated hold, redial, and transfer keys are supplied to facilitate call handling in a retail environment. Illuminated mute and speakerphone keys are provided to give a clear indication of speaker status. A pixel-based display with a white backlight makes calling information easy to see, and Extensible



Markup Language (XML) services deliver a rich user experience. The Cisco Unified IP Phone 7931G offers numerous important security features plus the choice of IEEE 802.3af Power over Ethernet (PoE) or local power through an optional power adaptor. Currently, there is no add-on line/feature key module available for the Cisco Unified IP Phone 7931G.

Features

The Cisco Unified IP Phone 7931G is designed to grow with your organization. A dynamic, soft-key driven feature set allows the phone to keep pace with your requirements through regular software upgrades. You can easily move phones, add new phones, and change existing phone arrangements; users can simply pick up their phones and move to a new location anywhere on the network. The Cisco Unified IP Phone 7931G also provides accessibility features for those with special needs.

• Lighted line keys—Twenty-four lighted line keys to which can be assigned individual lines-Each line key provides a busy-line indication if the line is shared with another Unified IP Phone; lighted line keys are also used to access services and call history directories and to activate the headset port.

• Dedicated keys for hold, redial, and transfer—The hold key is colored red to make it clearly visible in a fast-moving call environment; the redial and transfer keys are provided to facilitate rapid call handling.

• Lighted message waiting indicator —Lights turn on when there is new voicemail and when the phone rings; visible on both the phone chassis and handset and stay lit until new voicemail has been processed by the user.

• Graphical display—Graphical monochrome display with resolution of 192 x 64 pixels and a white backlight provides scrollable three-line intuitive access to calling features and text-based XML applications; the Cisco Unified IP Phone 7931G also supports audio-based XML applications.

• Four soft keys and a four-way rocker key—Dynamically present calling options to the user; the four-way rocker key allows easy movement through the displayed information.

• Network features—Cisco Discovery Protocol; IEEE 802.1 p/q tagging and switching.

• Ethernet switch —10/100BASE-T Ethernet connection through two RJ-45 ports: one for the LAN connection and the other for connecting a downstream Ethernet device such as a PC.

4.2.3 Professional Desktop IP Telephone Instrument The Professional model will be used by VoiceCon managers. It should have, at minimum the following design attributes and features/functions:

• 12 key dial pad • Six (6) programmable line/feature keys with soft label/status

indicators



• G711, G729 and wideband voice codecs • Auto Self Discovery/DHCP • Echo Canceller • QoS Support (802.1p/Q, DiffServ) • Hold key • Last Number Redial key • Release key • Message Waiting/Call Ringing indicator(s) • Full Duplex Speakerphone • Speaker/Mute key • Volume Control keys/slide • High resolution, backlit, monochrome grayscale pixel-based, graphical

display screen with four (4) associated context sensitive soft feature labels ((key, cursor, or navigator control)


numbers) • Integrated Ethernet switch and two (2) RJ-45 connector interface ports

for 10/100 Mbps connectivity • Bluetooth interface for wireless headset • USB interface • IEEE 802.af POE support

The Professional model must also be capable of supporting the following integrated feature/functions if required at some future time:

• Gigabit (10/100/1000 Mbps) Ethernet connectivity • Embedded Web-browser applications

Vendor Response Requirement: Confirm that your proposed Professional model satisfies the stated requirements and provide a brief product description that includes an illustration or photograph (PPT format, only) of the instrument. Indicate in your response any and all requirements not satisfied. State which required feature-specific keys are not available, but softkey feature access can be used as an alternative.




Cisco Response: The Cisco Unified IP Phone 7961G has been quoted to meet this requirement. Cisco Unified IP Phones provide unmatched levels of integrated business functionality and converged communications features beyond today's conventional voice systems and surpassing competitive offerings. As the market leader in IP Telephony, Cisco continues to deliver unparalleled end-to-end data and true voice-over-IP (VoIP) solutions, offering the most complete, stylish, fully featured Unified IP Phone portfolio to enterprise and small-to-medium sized customers.

The Cisco Unified IP Phone 7961G, a key offering in the Unified IP Phone portfolio, is a fully-featured Unified IP Phone perfect for managers and administrative assistants. It provides six programmable line and feature buttons, and a high quality speakerphone, plus enhanced functionality for those customers requiring additional capabilities. The new functionality and features are integrated within an industry-proven, award-winning industrial design.

This state-of-the-art Unified IP Phone includes a high-resolution, 4-bit grayscale display (320 x 222) for easy access to communication information, timesaving applications, and feature usage. It also enables customers and developers to deliver more innovative and productivity-enhancing Extensible Markup Language (XML) applications to the display. Dynamic back-lit tri-color buttons provide straightforward call state identification. Both Cisco pre-standard Power over Ethernet (PoE) and IEEE 802.3af PoE are supported.

The Cisco Unified IP Phone 7961G does not support Bluetooth or have a USB interface.

Features

The Cisco Unified IP Phone 7961G is dynamic and designed to grow with system capabilities. Features will keep pace with new changes via software updates to the phone’s flash memory. The phone provides many accessibility methods according to user preference.

The Cisco Unified IP Phone 7961G does not require hands-on moves or changes. Users can simply pick up the phone and move to a new location anywhere on their network – without the help of a system administrator.

Messages -- The message key provides direct access to voice mail.

Directories -- The Cisco Unified IP Phone 7961G identifies incoming messages and categorizes them on the screen. This allows users to quickly and effectively return calls using direct dial-back capability. The corporate directory integrates with the Lightweight Directory Access Protocol (LDAP3) standard directory.

Settings -- The settings feature key allows the user to adjust display contrast and select from a large number of ringer sounds and volume settings for all audio such as ringer, handset, headset, and speaker. Network Configuration preferences can also be set up (usually by the



system administrator). Configuration can either be automatic or manually set up for Dynamic Host Control Protocol (DHCP), Trivial File Transfer Protocol (TFTP), Unified CallManager software, and backup Unified CallManager software.

Services -- The Cisco Unified IP Phone 7961G allows users to quickly access diverse information such as weather, stocks, quote of the day, or any Web-based information. The phone uses XML to provide a portal to an ever-growing world of features and information.

Help -- The online help feature gives users information about the phone's keys, buttons, and features. The pixel display allows for greater flexibility of features and significantly expands the information viewed when using features such as Services, Information, Messages, and Directory. For example, the Directory button can show local and server-based directory information.

Volume Control, Microphone Mute Button, and Speaker On/Off Button Speakerphone -- The Cisco Unified IP Phone 7961G features high-quality speakerphone technology. It also includes an easy-to-use speaker on/off button and microphone mute buttons. These buttons are lit when active. The convenient volume control button provides for easy decibel-level adjustments for the speakerphone, handset, headset, and ringer.

Additional features

• The internal Cisco 2-port Ethernet switch allows for direct connections to a 10/100BASE-T Ethernet network via an RJ-45 interface with single LAN connectivity for both the phone and a co-located PC. The system administrator can designate separate virtual LANs (VLANs) (802.1Q) for the PC and Cisco Unified IP Phones, providing improved security and reliability of voice and data traffic.

• A dedicated headset port eliminates the need for a separate amplifier when using a headset. This allows the handset to remain in its cradle, making headset use simpler.

• The handset is hearing aid compatible (HAC) and meets Federal Communications Commission (FCC) loudness requirements for the Americans with Disabilities Act (ADA). Section 508 loudness requirements can be achieved using industry standard inline handset amplifiers such as Walker Equipment W-10 or CE-100 amplifiers.

• The dial pad is also ADA-compliant. • The foot-stand of the Cisco Unified IP Phone 7961G is adjustable from

flat to 60 degrees to provide optimum display viewing and comfortable use of all buttons and keys. The foot-stand is keyed to match standard wall jack configurations for wall mounting. Two optional wall mount brackets are also offered as noted below.

• For added information security, the audible dual-tone multi-frequency (DTMF) tones are masked when the speakerphone mode is used.

• The Cisco Unified IP Phone 7961G supports up to two Cisco Unified IP Phone Expansion Module 7914 units.



Other Cisco Unified IP Phone 7961G features include:

• 24+ user-adjustable ring tones

• G.711 and G.729a audio compression

• An IP address assignment - DHCP client or statically configured

• Comfort noise generation and voice activity detection (VAD) programming on a system basis

The phone also includes the following adjustable settings:

• User Preferences: Ring Tones, Background Images and Contrast

• Network Configuration

• Device Configuration

• Security Configuration

4.2.4 Executive Desktop IP Telephone Instrument The Professional model will be used by VoiceCon’s executive management team. It should have, at minimum the following design attributes and features/functions:

• 12 key dial pad • Twelve (12) programmable line/feature keys with soft label/ status

indicators • G711, G729 and wideband voice codecs • Auto Self Discovery/DHCP • Echo Canceller • QoS Support (802.1p/Q, DiffServ) • Hold key • Last Number Redial key • Release key • Message Waiting/Call Ringing indicator(s) • Full Duplex Speakerphone • Speaker/Mute key • Volume Control keys/slide • High resolution, backlit, color pixel-based, graphical display screen

with four (4) associated context sensitive soft feature lablels (key, cursor, or navigator control)


numbers) • Integrated Ethernet switch and two (2) RJ-45 connector interface

ports; 10/100 Mbps connectivity • Headset interface (Bluetooth is also acceptable) • IEEE 802.af POE support

The Professional model must also be capable of supporting the following integrated feature/functions if required at some future time:



• Gigabit (10/100/1000 Mbps) Ethernet connectivity • Embedded Web-browser applications

Vendor Response Requirement: Confirm that your proposed Executive model satisfies the stated requirements and provide a brief product description that includes an illustration or photograph (PPT format, only) of the instrument. Indicate in your response any and all requirements not satisfied. State which required feature-specific keys are not available, but softkey feature access can be used as an alternative.


Cisco Response: The Cisco Unified IP Phone 7970G has been quoted to meet this requirement. Cisco Unified IP Phones provide unmatched levels of integrated business functionality and converged communications features beyond today's conventional voice systems and surpassing competitive offerings. As the market leader in IP Telephony, Cisco continues to deliver unparalleled end-to-end data and true voice-over-IP (VoIP) solutions, offering the most complete, stylish, fully featured Unified IP Phone portfolio to enterprise and small-to-medium sized customers.

The Cisco Unified IP Phone 7970G is truly a prestige set, demonstrating the latest technology and advancements in VoIP telephony. It addresses not only the needs of the executive or major decision-maker but also brings network data and applications to users without PCs. This state-of-the-art Unified IP Phone includes a backlit, high-resolution color touch-screen display (320 x 234, 12 bit display with 4,096 colors) for easy access to communication information, timesaving applications, and feature usage. It also enables customers and developers to deliver more innovative and productivity-enhancing Extensible Markup Language (XML) applications to the display. Access to eight telephone lines (or combination of lines and direct access to telephony features), a high quality hands-free speakerphone, a built-in headset connection, and both Cisco pre-standard Power over Ethernet (PoE) and IEEE 802.3af PoE are supported. It supports 8 line keys.

Features The Cisco Unified IP Phone 7970G is dynamic and designed to grow with system capabilities. Features will keep pace with new changes via software



updates to the phone’s flash memory. The phone provides many accessibility methods according to user preference.

The Cisco Unified IP Phone 7970G does not require hands-on moves or changes. Users can simply pick up the phone and move to a new location anywhere on their network – without the help of a system administrator.

With both a color display and touch screen, the Cisco Unified IP Phone 7970G delivers more powerful applications and network data to the desktop.

Messages— The message key provides direct access to voice mail.

Directories — The Cisco Unified IP Phone 7970G identifies incoming messages and categorizes them on the screen. This allows users to quickly and effectively return calls using direct dial-back capability. The corporate directory integrates with the Lightweight Directory Access Protocol (LDAP3) standard directory.

Settings — The settings feature key allows the user to adjust display contrast and select from a large number of ringer sounds and volume settings for all audio such as ringer, handset, headset, and speaker. Network Configuration preferences can also be set up (usually by the system administrator). Configuration can either be automatic or manually set up for Dynamic Host Control Protocol (DHCP), Trivial File Transfer Protocol (TFTP), Unified CallManager software, and backup Unified CallManager software.

Services — The Cisco Unified IP Phone 7970G allows users to quickly access diverse information such as weather, stocks, quote of the day, or any Web-based information. The phone uses XML to provide a portal to an ever-growing world of features and information.

Help — The online help feature gives users information about the phone's keys, buttons, and features. The pixel display allows for greater flexibility of features and significantly expands the information viewed when using features such as Services, Information, Messages, and Directory. For example, the Directory button can show local and server-based directory information.

Volume Control, Microphone Mute Button, and Speaker On/Off Button Speakerphone — The Cisco Unified IP Phone 7970G features high-quality speakerphone technology. It also includes an easy-to-use speaker on/off button and microphone mute buttons. These buttons are lit when active. The convenient volume control button provides for easy decibel-level adjustments for the speakerphone, handset, headset, and ringer.

Stereo Jack Sockets — Located on the side of the Cisco Unified IP Phone 7970G is a 3.5mm stereo jack socket for connection to PC style speakers or headphones, and a second 3.5mm stereo jack socket for connection to a stereo microphone.



Note: These currently ship from the factory with a cover over the sockets. ONLY approved auxiliary speakers, headphones, or microphones can be used. Field notice will be sent at a later when these are available. An external power adapter will be required to drive these accessories.

Display Key — The display key provides easy access to previous "pages" or applications still open on the LCD.

Additional features

• The internal Cisco 2-port Ethernet switch allows for direct connections to a 10/100BASE-T Ethernet network via an RJ-45 interface with single LAN connectivity for both the phone and a colocated PC. The system administrator can designate separate virtual LANs (VLANs) (802.1Q) for the PC and Cisco Unified IP Phones providing improved security and reliability of voice and data traffic.

• A dedicated headset port eliminates the need for a separate amplifier when using a headset. This allows the handset to remain in its cradle, making headset use simpler.

• The handset is hearing aid compatible (HAC) and meets Federal Communications Commission (FCC) loudness requirements for the Americans with Disabilities Act (ADA). Section 508 loudness requirements can be achieved using industry standard inline handset amplifiers such as Walker Equipment W-10 or CE-100 amplifiers.

• The dial pad is also ADA-compliant. • The foot-stand of the Cisco Unified IP Phone 7970G is adjustable from

flat to 60 degrees to provide optimum display viewing and comfortable use of all buttons and keys. The foot-stand is keyed to match standard wall jack configurations for wall mounting. Two optional wall mount brackets are also offered as noted below.

• For the Cisco Unified IP Phone 7970G to have full display brightness, the external power adapter is required. The Cisco Unified IP Phone 7970G can receive power down the LAN from any of the Cisco inline power-capable blades boxes; however, the display screen is "half bright" and therefore is not the recommended mode of operation.

• In addition, the Cisco Unified IP Phone 7970G supports both the Cisco pre-standard Power over Ethernet (PoE) and IEEE 802.3af PoE.

• For added information security, the audible dual-tone multi-frequency (DTMF) tones are masked when the speakerphone mode is used.

• The Cisco Unified IP Phone 7970G does not support the Cisco Unified IP Phone Expansion Module 7914 at this time. Field notice will be sent at a later when this is accessible.

Other Cisco Unified IP Phone 7970G features include:

• 24+user-adjustable ring tones

• G.711 and G.729a audio compression

• An IP address assignment-DHCP client or statically configured

• Comfort noise generation and voice activity detection (VAD) programming on a system basis



The phone also includes the following adjustable settings:

• Display contrast

• Ring type

• Network configuration

• Call status

4.2.5 Desktop IP Telephone Instrument Web-browser Functionality Vendor Response Requirement: Provide a brief description of embedded Web-browser functionality for the proposed Professional and Executive IP desktop telephone instrument models. Include the following information in your response: browser protocol (HTML, XML, WAP, Java, LDAP, Stimulus, other); station user interaction (touchscreen and/or keypad control cursor control; ability to place calls during active screen applications; screen saver option; standard and optional applications (visual mailbox; personal directory and calendar; web page access and display; visual alerts; audio alerts; et al).

Cisco Response: Cisco Unified IP phones support web-based applications using XML services on the phones and graphical interfaces. Information can be displayed for the user, and responded to via key-pad input, or touch screen input in the case of the Cisco Unified IP Phone 7970G and 7971G which have color, touch screen displays.

As organizations of all sizes become accustomed to the integrated management capabilities of IP-based networks, developers are discovering a small but burgeoning applications market for IP phones. There are at least 10 times as many telephones in the world as there are computers. Phones, unlike computers, are always on. IP phones combine the most crucial capabilities of phones, pagers, and computers, in that they can be used for signaling, voice communications, and data communications.

Applications developed for IP phones are especially effective in places where phones are more logical, convenient, or ubiquitous than traditional computers:

• An IP phone is less likely than a computer to need frequent upgrades or servicing to withstand the rigors of dust and vibration found on factory floors

• IP phones are less expensive to maintain than PCs • Using a phone for multiple purposes saves space

A variety of examples of innovative and interesting phone-based applications exist today. A few examples are outlined below. Additional information is available at the Cisco IP Communications Applications Central web site located at:



http://forums.cisco.com/eforum/servlet/IPCApps?page=main

Emergency Warnings: Tornado warnings are usually announced by the media, but those who need information sooner can buy decoder boxes that pick up National Weather Service (NWS) radio signals and begin flashing a red light in response. The 17-school Frederick County School District in Virginia had installed such decoders, but they were only effective if someone was in the office to see them. Cisco partner AAC created a system to transmit the signal to the phones: Now, NWS warnings are automatically delivered to the LCD screens and speakers on all the district's IP phones, even wireless units.

Security: This era of heightened security also makes IP-based phones appealing for public-address systems. The headquarters of the U.S. Department of Commerce uses Berbee InformaCast to make announcements through the speakers of the agency's more than 4,200 IP-based phones. In cases of evacuation orders, the announcement can be made quickly and simultaneously. In Charles County Public Schools in La Plata , Maryland , the phones interact with the Verizon phone system. A 911 call automatically sends a detailed campus map of the caller's location to police headquarters.

School Attendance Tracking: Most schools now have Ethernet connections throughout their campuses, so they can use IP-based phones to transmit text or graphics without a major infrastructure upgrade. Attendance applications such as CalAmp ExtendTime and AAC PhoneTop K-12 Application Suite can provide parents ' contact information, so if the school hasn't been alerted that an absent child is sick, staff can contact the parents more quickly. In an emergency, they can also provide medical information about the student and quick access to medical help or police.

Time-Card Tracking: Attendance tracking is also useful in industries that have workers who are paid hourly, are highly mobile, or both. Being able to "punch in" using a phone keypad, or by swiping an ID card through an attached device, has advantages over the traditional time-clock system or even applications on computers.

Other natural markets for IP phone applications include vertical applications, such as in the healthcare industry, where people often need fast access to information and phones are usually more easily accessible than computers.

4.2.5 Desktop Instrument Options and Add-on Modules Vendor Response Requirement: Provide a brief description of all hardware/software options and/or add-on modules currently available with the proposed Economy, Administrative, Professional, and Executive models. Options/modules should include key modules, display modules, BlueTooth interface, USB interface, Gigabit Ethernet connectors, et al. necessary to satisfy the above telephone model requirements. Indicate the specific models that support the individual



option/module.

Cisco Response: No add-on modules were quoted for this RFP. The only add-on module offered at present is the Cisco Unified IP Phone Expansion Module 7914 (described in Section 4.8 below).

4.2.6 SIP Compatibility It is desirable, but not required, that the proposed desktop IP telephone instruments conform to current SIP standards and specifications at time of installation and system cutover. If any or all the proposed instrument models do not support natively embedded SIP capabilities as proposed, then it is acceptable that a firmware download upgrade be available when requested. Vendor Response Requirement Indicate which of the proposed telephone models are native SIP or can be reprogrammed via a SIP firmware download when requested by VoiceCon. Identify which proposed models cannot currently be reprogrammed for SIP support at this time.

Cisco Response: The proposed Cisco Unified IP Phones support both Cisco Skinny Client Control Protocol and SIP. They can be ordered initially with either protocol, and can be converted at any time with a firmware download. Support for SIP in other Cisco Unified IP Phone models is detailed in the tables in Section 4.8 below.

4.3 PC Client Softphone A PC client softphone will be used by station users and attendant operators as their primary desktop voice terminal. It is desirable, but not mandatory, that the PC client softphone application conform to SIP standards and specifications. Vendor Response Requirement Indicate if the proposed softphone solution satisfies the stated SIP requirement. If the proposed softphone solution does not support SIP, does your product portfolio currently include a PC client softphone solution that does?



IP Communicator

Cisco Response: Cisco® Unified IP Phones provide unmatched levels of integrated business capabilities and converged communications features that go beyond today's conventional voice systems and surpass competitive offerings as well. Cisco Systems® delivers unparalleled end-to-end data and IP Telephony solutions, offering the most complete, full-featured Unified IP Phone portfolio to enterprise and small- and midsized-business customers.

Cisco IP Communicator—a software-based application that delivers enhanced telephony support through personal computers—features the latest technology and advancements available with VoIP today. This application endows computers with the functionality of Unified IP Phones, providing high-quality voice calls on the road, in the office, or from wherever users may have access to the corporate network.

Cisco IP Communicator is designed to meet diverse customer needs as a supplemental telephone when traveling, a telecommuting device, or a primary desktop telephone. When using Cisco IP Communicator remotely, users aren't just taking their office extension with them, they also have access to the same familiar phone services they have in the office.

Cisco IP Communicator uses Cisco Unified CallManager call processing system to provide advanced telephony features and VoIP capabilities. When registered to Cisco Unified CallManager system, Cisco IP Communicator has the features and functionality of a full-featured Cisco Unified IP Phone, including the ability to transfer calls, forward calls, and conference additional participants to an existing call. This also means that system administrators



can provision Cisco IP Communicator as they would any other Cisco Unified IP Phone, greatly simplifying Unified IP Phone management.

This solution also enables customers and developers to deliver more innovative and productivity-enhancing Extensible Markup Language (XML)-based applications to the display. Access to eight telephone lines (or a combination of lines and direct access to telephony features) is included.

Cisco IP Communicator is a dynamic solution that is designed to grow with new system capabilities. Features will keep pace with new changes via automatic software updates.

Basic Features The Cisco IP Communicator includes all the features of a desktop business telephone.

• Feature parity with Cisco’s advanced Unified IP Phones

o Multiple lines o Unified IP Phone services support (XML) o Integrates with CTI or Unified IP Phone services application written

by Cisco or its partners o Localization

• SCCP-based to work with Cisco Unified CallManager 3.3 and 4.0 o All call control features like shared lines, park, pickup, meet me

conference, etc. o Same manageability and scalability as Cisco’s Unified IP Phones

• Multiple display options (skins) and quick launching • Windows 2000 and Windows XP support • Premium audio quality

o Kernel mode operation reduces the effect of resource intensive applications on audio quality

o Advanced jitter buffer and packet loss concealment algorithms o Audio "Tuning Wizard“ for setting input and output levels o Echo suppression and noise cancellation o Full-duplex speaker phone mode o IP precedence marking (layer 3) for audio priority

• VPN support o Auto-detection of Cisco VPN client o Automated support for most non-Cisco VPN clients

• Full USB telephony device support (HID support) • Unified administration with Cisco Unified IP Phones • Automatic software updates for deployment and management ease • Cisco Unified CallManager redundancy support • Ability to manage user configurations • Survivable Remote Site Telephony (SRST) support • Survivable routing capabilities in the event of WAN outages • Cisco Discover Protocol (CDP) support for Cisco Emergency Responder

(E911) application 4.3.1 Desktop Station User Application



The proposed PC client softphone solution must be able to support a minimum of six programmable line appearances, integrated system and personal directories with search/dial-by-name capabilities, and functions comparable to the proposed Professional model. The softphone solution must also be able to support a peripheral headset. Vendor Response Requirement Confirm that the proposed softphone solution satisfies the stated requirements and provide a brief product description that includes a illustration/photograph (PPT format, only) that depicts the look and feel of an active call screen display.

Cisco Response: Cisco Complies. See information in Section 4.3 above. 4.3.1.1 Teleworker Station User Application The proposed PC client softphone solution may also be used by some station users as a teleworker voice terminal outside the VoiceCon facility environment. Vendor Response Requirement Confirm that the proposed customer premises softphone solution can be used as am off-premises teleworker voice terminal option . Indicate in your response if any optional hardware and/or software requirements are required to support teleworker mode operations for deployment in a home, hotel, or office environment.

Cisco Response: Cisco Complies. See information in Section 4.3 above. 4.3.2 Soft Attendant Console Attendant operator console requirements are to be satisfied using a PC client softphone application. The attendant console application should include several distinct display fields, such as: incoming call queue and active caller information; release loop keys; feature/function keys; direct station selection (contact directory)/ busy lamp field; trunk groups; minor/major alarms; and messaging. GUI capabilities must support drag & click operations. At minimum the following information and data must be available in the softphone screen display: # Calls in queue; Call appearance status; Calling/called party number/name; Trunk ID; COS/COR; # Calls waiting; call coverage status; time/date, call duration; text messages; alarm notification Vendor Response Requirement Confirm the proposed softphone solution satisfies the stated requirements, and provide a brief description of the proposed softphone solution when programmed for attendant console operation. Include in the response a



representative illustration or photograph (PPT format, only) that conveys the look and feel of an active call console display screen.

Cisco Response: Cisco has included the Cisco® Unified CallManager Attendant Console in this response. Cisco® Unified CallManager Attendant Console is an application from Cisco Systems® that supports the traditional role of a manual attendant-console hardware device. Associated with a Cisco Unified IP Phone, the application allows the attendant to quickly accept and dispatch calls to users. An integrated directory service provides traditional busy-lamp-field (BLF) and direct-station-select (DSS) functions for any line in the system.

Cisco Unified CallManager Attendant Console integrates traditional time-division multiplexing (TDM) telephony functions with advanced IP telephony applications and services such as Lightweight Directory Access Protocol (LDAP) directory. A primary benefit of Cisco Unified CallManager Attendant Console over traditional attendant-console systems is its ability to monitor the state of every line in the system and to efficiently dispatch calls. The absence of a hardware-based line-monitor device offers an affordable and mobile attendant solution.

IMPORTANT FEATURES AND BENEFITS Companies today can choose to route inbound telephone calls through numerous methods. These methods are either completely automated, manually directed, or some hybrid of automated and manual operation. A separate product, the Cisco Automated Attendant application can accept inbound calls, query the caller for destination information, and rapidly dispatch the call without operator intervention. Automation of inbound call dispatch is efficient and affordable. Alternatively, many businesses see the benefit of handling each inbound caller through a specially trained and equipped operator. This operator assesses the caller's purpose and intended destination and uses tools to dispatch the call reliably and efficiently. The benefit of such a function is a heightened sense of customer satisfaction and, in many cases, a more reliably dispatched call. The Cisco Unified CallManager Attendant Console is designed to more efficiently automate both the user operations and the administrative operations of a manual attendant function. The attendant console uses an intuitive and configurable graphical user interface as the primary means of call handling and line-state monitoring. The software nature of the attendant console allows assignment of line-state monitors without the need of physically relabeling extender boxes with each line-monitor change. The directory pane displays the results of queries into the directory of all users in the system. The line-state and call-forwarding status of each user's primary line is presented with each record entry. The benefit over traditional consoles and line extenders is that each user's line is monitored, as opposed to monitoring only a select few in a TDM-based system (refer to the figure below).



The Cisco Unified CallManager Attendant Console Uses an Intuitive and Configurable Graphical User Interface as the Primary Means of Call Handling and Line-State Monitoring.

Advanced drag-and-drop capabilities and access to corporate LDAP directories combine to offer critical advantages over traditional manual attendant stations. In a system with hundreds or thousands of users, the attendant-console operator can accept calls and perform directory lookup by selecting the field title in the directory section and typing in the first few characters of the user's last name, first name, or department. A directory search that matches the query is returned. The operator can view the status of the user's line (forwarded, busy, idle, or ringing) and advise the caller of the line state. The operator can then transfer the call to the user by either initiating a traditional transfer sequence through the transfer function key or dragging and dropping the call from the selected loop to the desired user's record. The primary benefit of this user interface is quicker transaction time and subsequent customer satisfaction improvement. Even non-operators can benefit from the line-monitoring features of the Cisco Unified CallManager Attendant Console; by configuring the user interface to show only the speed-dials pane, regular phone users can monitor the state of their peers' phones and be more aware of their availability (See figure below).

The Cisco Unified CallManager Attendant Console User Interface Is Flexible Enough To Be Used as a Small Line-Status Tool



The Cisco Unified CallManager Attendant Console is scalable. Call-distribution groups can be assigned to any pilot number, which can in turn be assigned to one or more attendant-console loops. These loops represent answerable lines in a multiple attendant system. Calls are queued to one or more online attendant loops, thereby allowing scale and distribution among multiple operators. Multiple attendant consoles can be configured to monitor the same lines, affording scale to multiple operators when conditions require. Equivalent functions on a traditional system would require the purchase and administration of a line-extender device for each operator. Access to directory services and content extends the attendant toolbox for providing efficient, courteous service well beyond the capabilities of equivalent, traditional manual attendant functions.

NEW FEATURES IN VERSION 1.4 • Enhanced keyboard shortcuts to simplify navigation of user-interface elements; full mouseless operation is now possible

• Audible alerts provided on various call-related events

• Auto Hold option for invoking transfer, consult transfer, or conference

USER FEATURES • Loop keys (simultaneous management of all lines available on associated phone)

• Line states-idle, active, ringing, and unknown

• User label per line-monitor key for easy reference to user

• Per-call drag-and-drop transfer and hold-drag call from loop key to line-monitor key or directory record for transfer

• Per-call hold timers with visual and audible indicator that changes over time from green to yellow to red

• Headset capabilities of Cisco Unified IP Phones

• Answer and release

• Direct transfer

• Call join



• Log on and log off

• Feature to make attendant busy or available

• System supplementary features-hold, resume, transfer, consult transfer, park, conference, call waiting, and interposition call transfer

• Support for shared lines

• Support for more than two calls per line

• Extended audible alert on call presentation

• Single-button direct transfer to destination user's voicemail

• Display of all calls parked by any operator in a cluster

• Manual retrieval of parked call from display

• Display of all calls in broadcast hunt group with audible notification of new calls

• Manual retrieval of calls in broadcast hunt group

• Configurable keyboard shortcuts for alternative to mouse operation

• User-interface internationalization and localization to all Cisco Unified CallManager-supported languages

Speed-Dial View • Unlimited speed-dial keys with line monitoring

• Key grouping in multiple tabs

• Optional Notes field for more labeling options

Directory View • Line state-One record for every line appearance in the Cisco Unified CallManager cluster

• Query-Searchable by any field in the directory

• Sort-By last name, first name, extension, or department

• Call-forwarding status-Icon indicator of call forwarding of user's line to voicemail or another number for up to 10 entries in a directory search result

ADMINISTRATIVE FEATURES • Remote system or device installation and configuration through a Web browser

• Simultaneous line monitor by multiple operators-operators can view line state of any line from their console user interface

• Call distribution from a single pilot number to multiple directory numbers or user-line pairs

• Simultaneous monitoring of inbound calls from multiple operator positions

• Creation of up to 500 pilot numbers or distribution groups

SYSTEM CAPABILITIES • Availability-Provision for multiple operators on same loop or pilot and monitoring same line; if operator station fails or is off line, calls are distributed to all other operators with same loops

• Manageability-System device configuration through Cisco Unified CallManager Administration Web Interface



• Affordability-No line-extender hardware devices

SCALABILITY • Up to 5000 total hunt-group members divided in as many as 500 hunt groups (pilot numbers) per Cisco Unified CallManager cluster

• Four hunting algorithms: Longest Idle, First Available, Circular, and Broadcast

• Ability to queue up to 32 calls (on hold) to a pilot point

• As many call loops per attendant console as lines configured on the controlled Unified IP Phone device-Any loop can be assignable as a hunt-group member

• Five hundred Cisco Unified CallManager Attendant Consoles per cluster

• Busy hour call attempts (BHCAs) of 57 per attendant

4.4 IP Audio Conferencing Unit VoiceCon requires a limited number of desktop audio conferencing units with multidirectional, full duplex speakerphone operation. The unit must be native IP. Vendor Response Requirement Provide a brief description of the proposed IP audio conferencing unit and include in the response an illustration or photograph (PPT format, only) of the unit.

Cisco Unified IP Conference Station 7936 Cisco Response: The Cisco Unified IP Conference Station 7936 couples state-of-the-art conference room speaker-phone technologies with the Cisco award-winning voice communication technologies. The net result is a conference room phone that offers superior voice and microphone quality, with simplified wiring and administrative cost benefits. The Cisco Unified IP Conference Station 7936 is a full-featured, IP-based, high-quality hands-free conference station for use on desktops, conference rooms, and in executive suites. The Cisco Unified IP Conference Station 7936 easily attaches a Catalyst® 10/100 Ethernet switch port with a simple RJ-45 connection, and configures



itself to the IP network via the Dynamic Host Control Protocol (DHCP). The Cisco Unified IP Conference Station 7936 offers superior voice quality, virtually eliminating echoes, clipped words, and reverberations for more natural conversation. It features superior sound quality with a digitally tuned speaker and three gated microphones that minimize extraneous background noise, allowing conference participants to move around while speaking. Connecting the optional extension microphone kit to the base unit enables both voice coverage for larger rooms and enhanced speaker volume output. In addition to the regular telephony keypad, the Cisco Unified IP Conference Station 7936 provides three soft keys and menu navigation keys that guide a user through call features and functions. The Cisco Unified IP Conference Station 7936 also features a pixel-based LCD display with backlighting. The display provides features such as date and time, calling party name, calling party number, digits dialed, and feature and line status. Features

• Standard business telephony features - Call hold, call transfer, call release, mute, conference (ad-hoc and meet-me), park, pick up.

• Feature updates - Software upgrades from Cisco Unified CallManager will allow the product to grow along with system capabilities.

• Full-duplex operation - State-of-the-art acoustic technology permits natural, two-way conversations without clipping or distortion; the system automatically adapts to changes in the acoustic conditions of the room.

• Integrated keypad - Simplifies operation by eliminating the need to receive and place calls on a separate telephone.

• 360-degree room coverage - A powerful, digitally tuned custom speaker and three sensitive microphones provide uniform coverage of small to medium-sized conference rooms or offices.

• Single cable design - A single cable from the power interface module (PIM) cable combines network and power to reduce clutter on the tabletop.

• Simple to install - Configures with Cisco Unified CallManager. With automatic failover.

• No special end-user training required - Works like a regular telephone. • DHCP for auto address configuration to the IP network. • Cisco Discovery Protocol for Cisco Unified IP Conference Station 7936

to Catalyst switch port discovery. • Powerful protocol for E911 services, phone tracking, and asset and

theft management. • Auto configuration of phone number, software images, and

personalized settings – Simplifies installation, reconfiguration, and future feature enhancements such as Web-browsing capabilities.

The Cisco IP Conference Station 7936 uses a single 10/100BaseTx Ethernet LAN connection to the network via an RJ-45 interface. The Cisco Unified IP Conference Station 7936 also features:

• Convenient volume control buttons • Five-user adjustable ring tones



• G.711 (A-law and mu-Law) and G.729a audio compression • IP address assignment - DHCP client or statically configured • Comfort noise generation and voice activity detection • Web and LCD-based configuration • Local 20-entry directory

4.8 Other IP Telephone Instruments Please provide a brief description of additional IP desktop telephone instrument models included in your portfolio other than the models used to satisfy the Economy, Administrative, Professional, and Executive requirements. Information should include, at minimum, fixed feature/function, number of programmable line/feature keys, display description (if applicable), type of speakerphone (if applicable), and any other information you deem vital. Include an illustration/photograph (PPT format, only) for each of these additional models. Cisco Response: The Cisco Unified IP Phone portfolio is detailed in the four tables that follow. In addition, a PowerPoint file has been included with additional pictures.



Model

Cisco Unified IP Phone

7902G Cisco Unified IP

Phone 7905G Cisco Unified IP

Phone 7911G Cisco Unified

IP Phone 7912G

Integral switch No No Yes Yes

Number of line keys 1 1 1 1

Display No Pixel based 192 x 64 (monochrome)

Pixel based 192 x 64 (monochrome)

Pixel based 192 x 64 (monochrome)

Programmable (soft) keys

No 4 4 4

Fixed feature keys 6 2 2 2 Advanced features none none none none

Handsfree No No (call monitoring)

No (call monitoring)

No (call monitoring)

Message waiting indication

Yes Yes Yes Yes

3rd party XML support No Yes Yes Yes Headset port No No No No

Signaling Protocol SCCP SCCP SCCP SCCP Other Protocols supported

H323, SIP SIP SIP

802.3AF No No Yes No DHCP Yes Yes Yes Yes 802.1p/q Yes Yes Yes Yes



Model Cisco Unified IP Phone 7920

Cisco 7936 Conference

Station

Cisco Unified IP

Phone 7940G

Cisco Unified IP Phone 7941G and

7941G-GE

Integral switch n/a No 10/100 10/100 (7941G);

10/100/1000 (7941G-GE)

Number of line keys 6 1 2 2 lighted line-keys

Display Pixel based (monochrome)

Pixel based Pixel based 145 X 80 (grey scale)

Pixel based 320 x 222 (grey scale)


2 3 4 (+2 speed dial / line) 4 (+2 speed dial / line)

Fixed feature keys 3 6 8 8 Advanced features WEP and IEEE

802.1X LEAP authentication,

QoS, USB support

conference phone

none higher-resolution, more infrastructure integration options

Handsfree Yes - via earpiece

Yes Yes Yes


Yes - visual display

No Yes Yes

3rd party XML support Yes No Yes Yes Headset port 2.5mm

headset jack no Yes Yes

Signaling Protocol SCCP No SCCP SCCP Other Protocols supported

MGCP, SIP SIP

802.3AF No No No Yes DHCP Yes Yes Yes Yes 802.1p/q Yes Yes Yes Yes



Model


7960G

Cisco Unified IP Phone 7961G and 7961G-GE


Cisco Unified IP Phone 7971G-GE

Integral switch 10/100 10/100 (7961G); 10/100/1000 (7961G-GE)

10/100 10/100/1000

Number of line keys 6 6 lighted line-keys 8 lighted line-keys

8 lighted line-keys

Display Pixel based 145 X 80 (grey scale)

Pixel based 320 x 222 (grey scale)

320 x 234, 12-bit color depth

320 x 234, 12-bit color depth


4 (+2 speed dial / line)




Fixed feature keys 8 8 8 8 Advanced features additional line

keys with 7914 module

higher-resolution, more infrastructure integration options

color touch-screen

color touch-screen

Handsfree Yes Yes Yes Yes Message waiting indication

Yes Yes Yes Yes

3rd party XML support Yes Yes Yes Yes Headset port Yes Yes Yes Yes Signaling Protocol SCCP SCCP SCCP SCCP Other Protocols supported

MGCP, SIP SIP SIP SIP

802.3AF No Yes Yes Yes DHCP Yes Yes Yes Yes 802.1p/q Yes Yes Yes Yes



Model


Cisco Unified Video Advantage

Cisco IP Communicator


Expansion Module 7914

Integral switch 10/100 N/A N/A N/A Number of line keys 1 Up to 8 -

associated with Unified IP Phone

8 14 per, max 28 w/796X, 797X series

Display 4SIF (704 x 480 pixels), 4CIF (704 x 576 pixels),

PC settings PC settings Pixel based (grey scale)


5 5 (+8 speed dial / line)


Fixed feature keys 9 8 8 Advanced features integrated video

Unified IP Phone software application

software application

expansion module

Handsfree Yes Yes - associated with Unified IP Phone

Yes


Yes Yes - associated with Unified IP Phone

Yes

3rd party XML support Yes Yes Yes Headset port Yes Yes - associated

with PC Yes - associated with PC

Signaling Protocol SCCP SCCP SCCP SCCP Other Protocols supported

SIP SIP

802.3AF Yes n/a n/a DHCP Yes Yes Yes 802.1p/q Yes n/a n/a



5.0 Call Processing Features The proposed communications system should have a robust list of call processing features supporting station user, attendant, and system operations.

Cisco Response: Responses are provided below in the tables to indicate which features are available in the Cisco IP Communications system. For the latest information please refer to Cisco Unified CallManager documentation posted at: www.cisco.com A complete listing of Cisco Unified CallManager features is also included in Appendix A.

5.1 Station User Features It is required that the proposed communications system support the following list of station user features. Definitions for most listed features may be found in PBX Systems for IP Telephony (2002), written by Allan Sulkin and published by McGraw-Hill Professional. Table 9 Station User Features STATION USER FEATURES Cisco Response

ADD-ON CONFERENCE (6 party or more) YES

AUTOMATIC CALLBACK YES

AUTOMATIC INTERCOM YES

BRIDGED CALL APPEARANCE YES

CALLBACK LAST INTERNAL CALLER YES

CALL COVERAGE (PROGRAMMED) - INTERNAL & EXTERNAL CALL PROGRAMMING YES

TIME OF DAY/DAY OF WEEK CALL PROGRAMMING YES

ANI/DNIS/CLID CALL PROGRAMMING YES

INTERNAL CALLER ID PROGRAMMING YES

CALL FORWARDING - ALL CALLS YES

CALL FORWARDING - BUSY/DON'T ANSWER YES

CALL FORWARDING - FOLLOW-ME YES

CALL FORWARDING - OFF-PREMISES YES

CALL FORWARDING: RINGING YES

CALL HOLD YES

CALL PARK YES CALL PICKUP - INDIVIDUAL CALL PICKUP - GROUP

YES

CALL TRANSFER YES

CALL WAITING YES

CONSECUTIVE SPEED DIALING YES

CONSULTATION HOLD YES

CUSTOMER STATION REARRANGEMENT YES



DIAL BY NAME YES

DISCRETE CALL OBSERVING NO

DISTINCTIVE RINGING YES

DO NOT DISTURB YES

ELAPSED CALL TIMER YES

EMERGENCY ACCESS TO ATTENDANT YES

EXECUTIVE ACCESS OVERRIDE YES

EXECUTIVE BUSY OVERRIDE YES

FACILITY BUSY INDICATION YES

GROUP LISTENING NO

HANDS-FREE DIALING YES

HANDS-FREE ANSWER INTERCOM YES

HELP INFORMATION ACCESS YES

HOT LINE YES

INCOMING CALL DISPLAY YES

INDIVIDUAL ATTENDANT ACCESS YES

INTERCOM DIAL YES

LAST NUMBER REDIALED YES

LINE LOCKOUT YES

LOUDSPEAKER PAGING ACCESS YES

MALICIOUS CALL TRACE YES

MANUAL INTERCOM YES

MANUAL ORIGINATING LINE SERVICE YES

MEET ME CONFERENCING (6-Party or more) YES

MESSAGE WAITING ACTIVATION YES MULTI-PARTY ASSISTED CONFERENCE w/SELECTIVE CALL DROP MUSIC ON HOLD

YES

OFF-HOOK ALARM YES

PADLOCK YES

PAGING/CODE CALL ACCESS YES

PERSONAL CO LINE (PRIVATE LINE) YES

PERSONAL SPEED DIALING YES

PERSONALIZED RINGING YES

PRIORITY CALLING YES

PRIVACY - ATTENDANT LOCKOUT YES

PRIVACY - MANUAL EXCLUSION YES

RECALL SIGNALING YES

RINGER CUT-OFF NO

RINGING TONE CONTROL YES

SAVE AND REDIAL YES

SECONDARY EXTENSION FEATURE ACTIVATION NO SEND ALL CALLS SILENT MONITORING

YES

STEP CALL NO

STORE/REDIAL YES

SUPERVISOR/ASSISTANT CALLING YES

SUPERVISOR/ASSISTANT SPEED DIAL YES

TEXT MESSAGES YES

TIMED QUEUE YES

TRUNK FLASH YES



TRUNK-TO-TRUNK CONNECTIONS YES

WHISPER PAGE NO

Vendor Response Requirement Confirm that the proposed communications system supports each of the above listed station user features. Identify any and all features that are not included as part of the standard call processing software generic package. Identify any and all of the listed features that require additional hardware and/or software, e.g., CTI application server, because they are not included as part of the standard generic software package.

Cisco Response: Cisco supported features are indicated directly in the table above. “YES” means that the feature is standard and is included in the proposal. “NO” means that the feature is not currently supported. Note: A detailed list of Cisco Unified CallManager features is included in Appendix A.

5.1.1 Additional Station User Features Vendor Response Requirement Provide a listing of proposed standard generic software station user features that are not included in Table 9 that VoiceCon may find of use and benefit. 5.2 Attendant Operator Features It is required that the proposed communications system support the following list of attendant operator features. Definitions for most listed features may be found in PBX Systems for IP Telephony (2002), written by Allan Sulkin and published by McGraw-Hill Professional. Table 10 Attendant Operator Features

ATTENDANT OPERATOR FEATURES Cisco Response

AUTO-MANUAL SPLITTING YES

AUTO-START/DON'T SPLIT NO

BACK-UP ALERTING NO

BUSY VERIFICATION OF TERMINALS/TRUNKS NO

CALL WAITING YES

CAMP-ON NO

CONFERENCE YES

CONTROL OF TRUNK GROUP ACCESS NO

DELAY ANNOUNCEMENT NO

DIRECT STATION SELECTION w/BLF YES

DIRECT TRUNK GROUP SELECTION NO

DISPLAY YES

INTERCEPT TREATMENT YES



INTERPOSITION CALL & TRANSFER YES

INTRUSION (BARGE-IN) YES

OVERFLOW YES

OVERRIDE OF DIVERSION FEATURES NO

PAGING/CODE CALL ACCESS YES

PRIORITY QUEUE YES

RECALL NO

RELEASE LOOP OPERATION YES

SERIAL OPERATION YES

STRAIGHT FORWARD OUTWARD COMPLETION NO

THROUGH DIALING YES

TRUNK-TO-TRUNK TRANSFER YES

TRUNK GROUP BUSY/WARNING INDICATOR NO

TRUNK ID NO

Vendor Response Requirement Confirm that the proposed communications system supports each of the above listed attendant operator features. Identify any and all features that are not included as part of the proposed standard generic software feature package. Identify any and all features that require additional hardware and/or software, e.g., CTI application server, not standard with the proposed system model(s).


5.2.1 Additional Attendant Operator Features Vendor Response Requirement Provide a listing of proposed standard generic software attendant operator features that are not included in Table 10 that VoiceCon may find of use and benefit. 5.3 System Features It is required that the proposed communications system support the following list of system features. Definitions for most listed features may be found in PBX Systems for IP Telephony (2002), written by Allan Sulkin and published by McGraw-Hill Professional. Table 11 System Features

SYSTEM FEATURES Cisco



Response

ACCOUNT CODES YES

ADMINISTERED CONNECTIONS YES

ANSWER DETECTION YES

AUTHORIZATION CODES YES

AUTOMATED ATTTENDANT YES

AUTOMATIC CALL DISTRIBUTION YES

AUTOMATIC ALTERNATE ROUTING YES

AUTOMATIC CAMP-ON NO

AUTOMATIC CIRCUIT ASSURANCE NO

AUTOMATIC NUMBER ID YES

AUTOMATIC RECALL NO

AUTOMATIC ROUTE SELECTION - BASIC YES

AUTOMATIC TRANSMISSION MEASUREMENT SYSTEM YES CALL-BY-CALL SERVICE SELECTION YES

CALL DETAIL RECORDING YES

CALL LOG YES

CENTRALIZED ATTENDANT SERVICE YES

CLASSES OF RESTRICTION (SPECIFY #) YES

CLASSES OF SERVICE (SPECIFY #) YES

CODE CALLING ACCESS YES

CONTROLLED PRIVATE CALLS YES

DELAYED RINGING NO

DIAL PLAN YES

DIALED NUMBER ID SERVICE YES

DIRECT DEPARTMENT CALLING YES

DIRECT INWARD DIALING YES

DID CALL WAITING YES

DIRECT INWARD SYSTEM ACCESS NO

DIRECT INWARD TERMINATION YES

DIRECT OUTWARD DIALING YES

E-911 SERVICE SUPPORT YES

EXTENDED TRUNK ACCESS YES

FACILITY RESTRICTION LEVELS YES

FACILITY TEST CALLS NO FIND ME- FOLLOW ME NO FORCED ENTRY ACCOUNT CODES YES HOTELING (/PERSONAL ROAMING) YES HOUSE PHONE YES

HUNTING YES

INTEGRATED SYSTEM DIRECTORY YES

LEAST COST ROUTING (Tariff-based, TOD/DOW) YES

MULTIPLE LISTED DIRECTORY NUMBERS YES

MUSIC ON HOLD YES

NIGHT SERVICE –FIXED YES

NIGHT SERVICE - PROGRAMMABLE YES

OFF-HOOK ALARM YES

OFF-PREMISES STATION (OPX) YES



OPEN SYSTEM SPEED DIAL YES

PASSWORD AGING NO

POWER FAILURE TRANSFER STATION YES

RECENT CHANGE HISTORY YES

RESTRICTION FEATURES YES

CONTROLLED YES

FULLY RESTRICTED YES

INWARD/OUTWARD YES

MISCELLANEOUS TERMINAL YES

MISCELLANEOUS TRUNK YES

TOLL/CODE YES

TRUNK YES

VOICE TERMINAL (IN/OUT) YES

ROUTE ADVANCE YES

SECURITY VIOLATION NOTIFICATION YES

SHARED TENANT SERVICE YES

SNMP SUPPORT YES

SYSTEM SPEED DIAL YES

SYSTEM STATUS REPORT YES

TIME OF DAY ROUTING YES

TIMED REMINDER NO

TRUNK ANSWER ANY STATION YES

TRUNK CALLBACK QUEUING NO

UNIFORM CALL DISTRIBUTION YES

UNIFORM DIAL PLAN YES

VIRTUAL EXTENSION YES

VOICE MESSAGE SYSTEM INTERFACE YES


5.3.1 Additional System Features Vendor Response Requirement Provide a listing of proposed standard generic software system features that are not included in Table 11 that VoiceCon may find of use and benefit.

Cisco Response: Cisco Unified CallManager 5.X supports a rich set of features which are listed in detail in Appendix A. A few important features that VoiceCon may find of use an benefit are described below:

Multiprotocol Support: Cisco Unified CallManager natively supports multiple call processing protocols, including MGCP, H.323, SCCP and SIP.



It supports coexistence of SCCP and SIP phones, allowing migration to SIP while protecting investments in existing devices. Video Telephony: Cisco introduced its IP Video Telephony solution in Cisco Unified CallManager Release 4.0. Video is fully integrated into Cisco Unified CallManager, and there are also many video endpoints available from Cisco and its strategic partners. Cisco Unified Video Advantage is just as easy to deploy, manage, and use as a Cisco Unified IP Phone. The Cisco IP Video Telephony solution consists of Cisco Unified CallManager; Cisco Unified Videoconferencing 3500 Series Multipoint Control Units (MCUs) for H.323, Session Initiation Protocol (SIP), and Skinny Client Control Protocol (SCCP) conference calls; Cisco Unified Videoconferencing 3500 Series H.320 Gateways; Cisco IOS H.323 Gatekeeper; Cisco Unified Video Advantage; Cisco IP Video Phone 7985; Sony and Tandberg SCCP endpoint solutions; and the existing range of H.323 or SIP-compliant products from partners such as Polycom, Tandberg, Sony, and others.

Computer Telephony Integration (CTI): Cisco Unified CallManager 5.X has built-in support for CTI applications using the Cisco CTI gateway, or JTAPI or TAPI integrations. Computer telephony integration (CTI) enables you to leverage computer-processing functions while making, receiving, and managing telephone calls. CTI applications allow you to perform such tasks as retrieving customer information from a database on the basis of information that caller ID provides. CTI applications can also enable you to use information that an interactive voice response (IVR) system captures, so the call can be routed to the appropriate customer service representative or so the information is provided to the individual who is receiving the call. Extension Mobility: The Extension Mobility feature allows users to configure any Cisco Unified IP Phone as their own, on a temporary basis, by logging in to that phone. Once a user logs in, the phone adopts the user individual user default device profile information, including line numbers, speed dials, services links, and other user-specific properties of a phone. For example, when user A occupies a desk and logs in to the phone, her directory number(s), services, speed dials, and other properties appear on that phone; but when user B uses the same desk at a different time, his information appears. The Extension Mobility feature dynamically configures a phone according to the current user.

RSVP: Cisco Unified CallManager 5 supports Resource Reservation Protocol (RSVP) agent capability. The RSVP agent on a Cisco router extends CAC capability beyond a hub-and-spoke topology within a cluster. Now a call can be routed directly between two locations without having to traverse the hub, allowing alternative network topologies and more efficient use of networks. iDivert Feature: Cisco Unified CallManager allows immediate diversion of an incoming call or an in-progress call to voicemail. In Cisco Unified CallManager 5.1, the feature has been enhanced to address transferred calls as well. Users now have the option to forward calls that have been transferred to them either to their own voicemail or to the voicemail of the original transferring party.



5.4 Mobility Features VoiceCon requires that the proposed IPTS support a variety of features and applications to support its mobile workforce. 5.4.1 Fixed Teleworking VoiceCon requires that its employees be able to use a PC client softphone outside the office environment using Internet or VPN access. Vendor Response Requirement Verify that a VoiceCon employee can use a PC client softphone to access the full set of HQ IPTS features and functions from a remote location using Internet or VPN access. Specify if there are known NAT or firewall transversal issues with this application.

Cisco Response: Cisco IP Communicator—a software-based application that delivers enhanced telephony support through personal computers—features the latest technology and advancements available with VoIP today. This application endows computers with the functionality of Unified IP Phones, providing high-quality voice calls on the road, in the office, or from wherever users may have access to the corporate network through VPN or the Internet. Cisco IP Communicator is described in detail in Section 4.6 above.

Cisco IP Communicator uses Cisco Unified CallManager call processing system to provide advanced telephony features and VoIP capabilities. When registered to Cisco Unified CallManager system, Cisco IP Communicator has the features and functionality of a full-featured Cisco Unified IP Phone, including the ability to transfer calls, forward calls, and conference additional participants to an existing call. This also means that system administrators can provision Cisco IP Communicator as they would any other Cisco Unified IP Phone, greatly simplifying Unified IP Phone management.

This solution also enables customers and developers to deliver more innovative and productivity-enhancing Extensible Markup Language (XML)-based applications to the display. Access to eight telephone lines (or a combination of lines and direct access to telephony features) is included.

There are no known issues with NAT traversal.

5.4.2 Cellular Extensions VoiceCon requires that its employees be able to utilize their cellular handsets to answer and place calls that are routed through the HQ IPTS. Vendor Response Requirement Verify that the proposed IPTS solution can support cellular handsets as system extensions, and briefly describe the IPTS option. Off-premises call



forwarding of calls directed to an IPTS extension is not satisfactory for this requirement. Specifically address the following: 5.4.2.1 Shared directory number with IP desktop telephone instrument or

PC client softphone for inbound and outbound calls 5.4.2.2 Access to IPTS call answering and calling features (identify specific

features available to cellular handset user) 5.4.2.3 Shared voice mailbox for desktop stations and cellular extensions 5.4.2.4 If available provide a picture or photograph in PowerPoint format

of an optional graphical user interface screenshot for use with a cellular handset

5.4.2.5 Indicate if the cellular extension feature is proprietary to a specific cellular carrier operator service

5.4.2.6 Identify any optional hardware/software required to support the cellular extension feature if not a proposed IPTS generic software feature

Cisco Response: Cisco MobilityManager is the application server software that delivers Mobile Connect, Mobile Voice Access, and other mobility services. Cisco Mobility Manager runs on the MCS-7815 server with Rel 1.1. Mobile Connect (MC) is the end-user mobility service that delivers single number, single voicemail box, and device mobility functionality. Cisco® MobilityManager makes it easy for enterprise workers to keep in touch with the business at hand, whether at their desks or mobile. It introduces Mobile Connect enterprise mobility services to extend the benefits of IP Communications to workers inside and outside the enterprise campus. An application server that integrates with Cisco Unified CallManager, Cisco MobilityManager intelligently manages, filters, routes, and places calls between a worker's Unified IP Phone and remote mobile phone. With Cisco MobilityManager, a worker can receive and place business calls from the devices most convenient for the task without interrupting the calls, whether in the office, in transit, or at a remote location. Cisco MobilityManager also helps enterprise IT and telecom managers better serve the communication needs of their mobile workers, while enabling them to leverage the enterprise IP Communications network resources available with Cisco Unified CallManager. Cisco MobilityManager is installed on the Cisco 7800 Series Media Convergence Server (MCS) appliances.

Cisco MobilityManager Deployment Diagram:



Cisco MobilityManager is optional and runs on an adjunct Cisco MCS server. It has not been included in this proposal, but it could be added very cost effectively. For budgetary purposes, here is a breakdown of the list price of Cisco MobilityManager to support 500 VoiceCon users: Description Quantity List Price Total Cisco MobilityManager Software 1 $1,995 $1,995 Cisco Mobile Connect User Licenses 500 $100 $50,000 Cisco MobilityManager server (MCS 7815-H1)

1 $4,000 $4,000

Grand Total $55,995

SOLUTIONS

Single Business Number Reach Cisco MobilityManager makes Mobile Connect services available to Cisco Unified CallManager users who would like to consolidate all their business calls with a single enterprise Unified IP Phone number and immediately connect wherever they happen to be working. Enterprise customers now need only a single phone number to reach enterprise workers, and the enterprise can provide more responsive service with no additional effort. For enterprise mobile workers, Cisco MobilityManager also reduces the burden of having to share their private mobile phone number and having to check for business calls in their mobile voice mail box.

Single Business Voice Mail If mobile workers are unable to answer Mobile Connect calls, they can rely on Cisco MobilityManager to store the unanswered calls in Cisco Unity, or other enterprise voice mail system. Workers can manage all voice mail using the single enterprise voice mail box.



Device Mobility Mobile phones are great when moving from location to location, but when a mobile worker arrives at the office, the mobile phone becomes less convenient. With the Mobile Connect services of Cisco MobilityManager, mobile workers can continue a call on their IP desk phone after they arrive at the office and take advantage or speakerphone or other Unified IP Phone services. Important calls can be continued without interruption, and workers can use the best available IP or mobile features for the specific time and place.

Cisco Mobile Voice Access Cisco MobilityManager makes all the major enterprise IP communications features available to workers while they are traveling. For example, an enterprise mobile worker who needs to call one of the enterprise's foreign offices while traveling can use the Cisco Mobile Voice Access line to place the call as if from the enterprise home office. The worker dials the Cisco Mobile Voice Access line from the mobile phone and places the call on the enterprise IPC network over a tie line. The connection is completed, and telecom costs are kept under control.

Web-Based System and User Administration Cisco MobilityManager offers flexible options to define and manager user profiles. Users can access the secure User Profile Web pages to enter mobile and other remote phone numbers and create filters that restrict the types of calls that are directed using Mobile Connect services. System administrators can use the secure Administration pages to determine how much control users will have over their profiles and make user profile changes when needed. Users enjoy the advantages of personal choice, while the enterprise retains control over resource use and is able to provide backup support.

MobilityManager Web-Based Administration

FEATURES



The Cisco Mobile Connect service helps mobile workers direct their inbound business calls to their Unified IP Phone number and initiate outbound business calls as if they were at their Unified IP Phone-all from the convenience of the mobile phone (or any remote phone destination). They can answer incoming calls on the desk phone or mobile phone, pickup calls between the desk phone and mobile phone without losing the connection, and originate enterprise calls from a mobile or other remote phone. To support Mobile Connect, Cisco MobilityManager software is shipped with an integrated suite of mobility application services, including Web-based system administration and user profile configuration utilities to create, access, and control the user profile information for each enterprise mobile worker. Cisco MobilityManager offers the following features:

• Simultaneous desktop ringing-Incoming calls ring simultaneously on the

user's IP desktop phone and mobile phone or phones. As soon as the user answers one phone, the unanswered phones automatically stop ringing. The user can choose the preferred phone to answer each time a call comes in.

• Desktop pickup-If a user initiates a call from a mobile phone, the call can be picked up on the users' desktop phone without losing the connection.

• Mobile call pickup-If a user initiates a call from the desktop phone, the call can be switched to the user's mobile phone without losing the connection. Based on the needs of the moment, users can take advantage of the reliability of the wired office phone or the convenience of the mobile phone.

• Security and privacy for Mobile Connect calls-During an active Mobile Connect call, the associated desktop Unified IP Phone is secured. Access to the call from the desktop is eliminated as soon as the cellular connection becomes active, precluding the possibility of an unauthorized person listening in on the call that is bridged to the cell phone.

• Cisco Mobile Voice Access-Users can initiate calls from a mobile phone as if the phone is a local enterprise IP PBX extension and take full advantage of local voice gateways and WAN trunking.

• Single enterprise voice mailbox-User can rely on their enterprise voice mail box as the single, consolidated voicemail box for all business, including calls to the desktop and mobile phone. Incoming callers have a predictable means of contacting employees and less time is needed for users to check multiple voice mail systems.

• Allowed and blocked call filters-Users can create a restricted list of caller phone numbers for which they definitely want to trigger simultaneously ringing on their desktop and mobile phones (allowed call filter) and also create a list of phone numbers that will not cause their mobile phone to ring when the desktop phone rings (blocked call filter). This assures that each user can receive critical calls, while preventing promulgation of unwanted or unnecessary calls.

• Caller identification-Caller ID is preserved and displayed on all calls. Users can take advantage of Mobile Connect with no loss of the original caller information (subject to mobile phone service provider capabilities).

• System administrator-controllable user profile access-User profile settings can be modified by system administrators through the secure Cisco MobilityManager Administration web pages and by users through the secure User Profile web pages. System administrators can determine how



much control users have over their profiles, thereby preserving their ability to balance IP telephony resources with user choice.

• Remote on/off control-Users can turn Mobile Connect features on or off from a mobile phone using the Cisco Mobile Voice Access application or from the user profile pages, assuring flexibility in how mobility is managed.

• Voice-based access with user identification and personal identification number protection-The Cisco Mobile Voice Access application is protected by user name and password.

• Call tracing-Detailed Mobile Connect calls are logged, providing information to help the enterprise optimize trunk usage and debug connection problems.

5.4.3 Fixed Mobile Convergence VoiceCon may be interested in implementing a Fixed Mobile Convergence (FMC) solution at some future date to increase station user productivity and performance. FMC supports seamless communications between a premises WLAN and a service provider cellular network using the same mobile communications device, with access to and implementation of IPTS features and functions. Vendor Response Requirement Briefly describe current efforts and activities to support a FMC solution using a dual mode 802.11/GSM mobile communications device behind the proposed IPTS. Include in the response estimated availability dates of the FMC solution, required WLAN equipment to support premises roaming capabilities, QoS, and security. Also identify the means to provide seamless handoff between the 802.11 WLAN and the cellular network for active calls.

Cisco Response: Cisco's FMC solution includes several componets to fully address the market needs. To support FMC the wireless infrastructure must support the necessary features to provide a high quality Voice over WLAN (VoWLAN) experience. Cisco's Unified Wireless Network infrastructure has been designed to support features that are a requirement for VoWLAN. Features that should be considered for a VoWLAN network include: * Fast Roaming over Layer 2 and Layer 3 * 802.11e and WMM support * TSPEC for Call Admission Control * Power saving options such as U-APSD Cisco has also written design guides specifically for VoWLAN deployments and worked with our training partners to develop training classes around this technology. Cisco is working with dual-mode handset vendors to bring FMC to market. In our new Unified CallManager version 6.0 we will incorporate the capability to handoff between the 802.11 WLAN and the cellular network, and the ability to



do enterprise features while in the GSM network. The Nokia solution will encompass the 'E' Series of phones including the E60, E61, and E70. These phones will have both GSM and 802.11b/g radios. No other solution components are required other than the phone, Cisco wireless infrastructure, and Cisco Unified CallManager. The Nokia phones are planned to support both WPA and WPA2, along with several EAP types for security. Additional information will be posted when available on the following url: www.cisco.com



6.0.0 Systems Management The proposed communications system must be administered, monitored, and maintained through operations organized into five functional areas: Fault, Configuration, Accounting, Performance and Security. All of the systems and devices in your proposed solution should attempt to provide comprehensive operations in each area. Operations for each area must be accessible through one interface regardless of the underlying system or device being managed. If a proxy server is used for intermediate operations, there must be at most one central database for each functional area. Systems or devices may be accessed individually if no proxy server is used. EXCEPTION: Optional call center solutions may provide its own set of FCAPS management operations separate from the general enterprise communications solution. Any supplied management applications must support decentralized access from any distributed PC client across the HQ LAN/WAN infrastructure and remote dial-up PC clients. It is also desirable for the applications to support a browser based user interface for intensive remote operations. Any supplied management applications may integrate information from the five functional areas at the presentation level. Vendor Response Requirement Confirm and verify that each functional area required to manage the proposed IPTS network is supported by a single, centrally located proxy server or, alternatively, each system or device supports a single API for a given functional area. Provide a brief description of the proposed management system, including its major hardware and software components. Specify if the proposed systems management server and software is available as a bundled offering, only, or if VoiceCon is responsible for providing its own server hardware to operate the software. If third party technology is used, please indicate which components are managing your solution in a vendor agnostic fashion.

Cisco Response: Cisco Unified CallManager is administered via a web browser. All Cisco Unified CallManager administration and serviceability tools, as well as applications such as, Unity Voice Mail and Unified Messaging, Cisco Emergency Responder, Cisco Conference Connection, IP Contact Center Express (Unified Contact Center Express), etc., provide browser based management. Any PC on the WAN/LAN with appropriate security can gain access to Unified CallManager administration regardless of whether the server is local or remote. Netscape 4.5 or Internet Explorer 5.0 or better is required to administer the database. There is no limit on the number of simultaneous



sessions (admin workstations logged on). If desired, limits can be set within Windows 2000. The entire cluster is driven from a single web page through the server designated as the “publisher”. (The other servers in the cluster are referred to as “subscribers”). The web page sends inputs to the Publisher server, and those changes are then replicated to all Subscriber servers in the cluster. If you have multiple Unified CallManager clusters in an Enterprise, each cluster must be managed separately. There is no single web page to manage multiple clusters.

6.0.1 System/Port Capacity Vendor Response Requirement Identify the maximum number of independent IPTS communications systems that can be supported by the proposed systems management server, and the maximum number of user ports that can be passively and actively supported.

Cisco Response: Each cluster can be managed through a single instance of a browser interface. There is no actual limit on how many Unified CallManager clusters, or systems that can be managed from a single system administration station.

6.0.2 Terminal Capacity Vendor Response Requirement Identify the maximum number of configurable and active PC client terminals that can be configured as part of the proposed management server system.

Cisco Response: Each cluster can be managed through a single instance of a browser interface. There is no actual limit on how many PCs or terminals that can be simultaneously logged on.

6.0.3 Support for Open Standards The proposed management system should provide support for open protocols, such as LDAP and SNMP. The proposed management system should use open encoding schemes, such as XML and HTML. Vendor Response Requirement Briefly discuss the open standards included in your proposed management system that supports administration, operations and maintenance services. Indicate if any protocols or encoding schemes are de facto standards or are being implemented publicly by other vendors.

Cisco Response: The Cisco management and serviceability tools support a variety of standards including SQL, AXL/SOAP, LDAP and SNMP. • The Cisco Unified CallManager configuration database is stored in a

standard SQL database. • AXL/SOAP provides a standards based interface that Cisco third party

developers can use to create applications that access information directly



from the Cisco Unified CallManager SQL database. • LDAP is used to store and reference critical information, such as

passwords. • SNMP is used to capture information that can be used to diagnose

problems in the system or individual devices. An SNMP agent can send traps that identify important system events to the network manager. The following list illustrates a few examples of Cisco Unified CallManager SNMP trap messages that can be sent to an NMS that is specified as a trap receiver:

• Cisco Unified CallManager failed • Phone failed • Phones status update • Gateway failed • Media resource list exhausted • Route list exhausted • Gateway layer 2 change

When an SNMP agent detects an alarm condition, it generates a trap (a notification message) that is sent to a configured IP address. You first configure the SNMP services in the Windows 2000 server, and then designate what traps Unified CallManager will send through the “Alarm Configuration” interface.

6.0.3 Security Features Unauthorized access to the communications system is a major concern. The ability to detect security problems is desirable beyond mechanisms to prevent security problems. Vendor Response Requirement Briefly describe the security features that are embedded in the proposed management system to prevent unauthorized access and operation. Specify if media encryption is used for command signaling transmissions. What, if any, Denial of Service (DoS) and user authentication mechanisms are supported for the systems management application?

Cisco Response: Unified CallManager utilizes password protection to prevent unauthorized access to the management database. Multiple levels of administration allow any number of users to be assigned to any number of groups. User groups are assigned to Functional Areas which grant read/write, read-only, or no access to the various sections of the Unified CallManager administration web pages. The users may be stored in an LDAP directory (either Microsoft Active Directory or NetScape iPlanet) or are stored locally within the Unified CallManager database if an LDAP server is not configured (not that Cisco Unity provides an Active Directory to which Unified CallManager can be integrated if the customer does not already have their own LDAP directory. Unified CallManager synchronizes with the LDAP directory and performs all authentications requests against that directory. Audit trails and logs are kept for all authentication attempts. The system can be configured to use LDAP over SSL to secure the



communications between Unified CallManager and the back-end LDAP server(s). HTTP access to the management web interfaces is done using HTTPs. Shell access to the CLI interface is done using SSH. Access to the various areas of the database via the AXL/SOAP API is done using HTTPs. SNMPv3 is supported for SNMP access.

Operating System Protection from DoS Attacks: Cisco Unified CallManager uses highly secure operating systems including hardened versions of Windows and Linux. In addition, Cisco bundles a non-managed version of our Cisco Security Agent (CSA) on every telephony application. CSA is an intrusion prevention that can very effectively block “day zero” attacks because it is behavioral-based rather than signature-based. Cisco does not provide, but recommends, that customers install and run anti-virus software. Additional information about protection from DoS attacks is contained in Section 1.6.2 above. L2/L3 Security: Cisco offers a broad selection of security specific products and models such as firewalls, network intrusion detection, host-base intrusion prevention, identity management (AAA), and security information management. Cisco, more than any other vendor, is continually and aggressively adding security features to existing routers, switches and other networking products. As the leading provider of IP Telephony products, Cisco is leading the charge to secure IPT and setting the benchmark by which all other vendors are being measured. No other vendor can provide the depth and breadth of security related features and products for building a secure end-to-end IP telephony and data communications network. Signaling and Media Encryption: Both of these capabilities were introduced with Unified CallManager Release 4.0 and are described in detail in Sections 1.6.1 and 1.6.3 above.

6.0.5 User Interface & Tools The management system should be operated using by GUI tools, formatted screens, pull down menus, valid entry choices, templates, batch processing & transactions scheduling, and database import/export. In general you should support a user interface set for each functional area: fault, Configuration, Performance and Security. The constituent users of each of these areas are distinct and your interface for each should optimize the experience for that constituent group. Management applications my integrate information from several management areas to enhance one functional area being managed.

Cisco Response: Cisco complies. 6.1.0 Administration Functions The proposed systems management solution must support: station user moves, adds, and changes; trunk group definitions and individual trunk circuit programming; voice terminal parameters; call restriction assignments; class of service definitions and assignments; password resets; customer profile database; ARS routing tables; group definitions and assignments; first digit tables; dial plan; feature access codes; paging/code call zone assignments.



Vendor Response Requirement Confirm the proposed systems management solution supports each of the listed administrative functions. Identify any functions not supported.

Cisco Response: Cisco complies. 6.1.1 Group Assignments The administration subsystem must support each of the following group definitions and assignments

• Abbreviated Dialing (System, Group, Enhanced)

• Hunt Groups

• Call Coverage Answer Groups

• Pickup Groups

• Intercom Groups

• Terminating Extension Groups

• Trunk Groups

Vendor Response Requirement Confirm administration support for each of the listed group definitions. List any and all groups not supported by the administration subsystem.

Cisco Response: Cisco complies. 6.1.2 Facilities Performance Management & Reports The management system must be able to collect, analyze, and provide reports for a variety of system operations.

Cisco Response: Cisco complies. Serviceability

Administrators can use the Cisco Unified CallManager Administration service tool to troubleshoot system problems. This web-based tool, Serviceability, provides the following services:

• Alarms--Saves alarms and events generated by Cisco Unified CallManager services for troubleshooting and provides alarm message definitions.

• Trace--Saves trace information generated by Cisco Unified CallManager services to various log files for troubleshooting. Administrators can configure, collect, and analyze trace information.

• Real-Time Monitoring Tool--Monitors real-time behavior of the components in a Cisco Unified CallManager cluster.



• Service Activation--Displays activation status of Cisco Unified CallManager services. Administrators use Service Activation to activate and deactivate services.

• Control Center--Displays status of Cisco Unified CallManager services. Administrators use Control Center to start and stop services.

• Quality Report Tool (QRT)--Provides voice quality and general problem-reporting tool for Cisco Unified IP Phones 7940 and 7960.

System Administrators access Serviceability from the Cisco Unified CallManager Administration window by choosing “Applications” from the menu bar. Installing the Cisco Unified CallManager software automatically installs Serviceability and makes it available.

The Cisco Unified CallManager Serviceability reporting tool, CDR Analysis and Reporting (CAR) provides the following functions:

• Multiple levels of users—Administrators who can generate system reports, and configure system parameters; managers who can generate reports for users and departments; users who can generate individual billing reports.

• Generate user reports—User reports include individual bills, department bills, top N by charge, top N by duration, top N by number of calls, CTI port enabled, and Cisco Unified IP Phone services.

• Generate system reports—System reports include QoS detail, QoS summary, QoS by gateway, QoS by call types, traffic summary, traffic summary with extensions, system overview, and CDR error.

• Generate device reports—Device reports include gateway detail, gateway summary, gateway utilization, route group utilization, route list utilization, route pattern utilization, conference bridge utilization, and voice mail utilization.

• CDR search—Searches the CDR database to verify the details of a call helping to track the progress and quality of leg of a call.

• System configuration—Administrators configure system parameters, report scheduler, database options, and error and event logs.

• Report configuration—Administrators configure base rate and duration for calls, factoring options, QoS values, and automatic report generation/alert.

6.2.1 Basic Trunk Usage and Traffic Trunk traffic records should be kept for all inbound and outbound calls, identifying the trunk group and trunk channel, time and duration of call. Vendor Response Requirement Confirm that the proposed facilities management system satisfies this requirement.

Cisco Response: Cisco complies. The CDR Analysis and Reporting Tool includes several reports that meet these requirements. These reports are the 1) Gateway Detail Report, 2) Gateway Summary Report, and 3) Gateway and



Route Utilization Report. Gateway Detail Report: Field Description Date The date when the call went through the gateway.

Orig. Time The time when the call went through the gateway.

Term. Time The time that the call terminated.

Duration(s) The duration, in seconds, that the call was connected. The duration specifies the difference between the Dest Connect and the Dest Disconnect times.

Orig The directory number from which the call was placed.

Dest The directory number to which the call was originally placed. If the call was not forwarded, this directory number should match the Final Destination number. If the call was forwarded, this field contains the original destination number of the call before it was forwarded.

Orig. Codec The codec type (compression or payload type) that the call originator used on its sending side during this call. This type may differ from the codec type used on its receiving side.

Dest. Codec The codec type (compression or payload type) that the destination used on its sending side during this call. This type may differ from the codec type used on its receiving side.

Orig. Device The device name of the device that placed the call. For incoming and tandem calls, this field specifies the device name of the gateway.

Dest Device The device name of the device that received the call. For outgoing and tandem calls, this field specifies the device name of a gateway. For conference calls, this field specifies the device name of the conference bridge.

Orig QoS Quality of service shows the voice-quality grade achieved for the calls.

Gateway Summary Report: Field Description Call Classification

Shows the total number of calls for each call classification.

Quality of Service

Shows a summary of the performance of the various gateways with the total number of calls for each voice-quality category. Good—QoS for these calls specifies the highest possible quality.

• Acceptable—QoS for these calls, although slightly degraded, still falls within an acceptable range.

• Fair—QoS for these calls, although degraded, still falls within



a usable range. • Poor—QoS for these calls was unsatisfactory. • NA—These calls did not match any criteria for the established

QoS categories.

Calls Shows the total number of calls for the particular call classification.

Duration(s) Shows the total number of duration for all the calls for the particular call classification.

Gateway and Route Utilization Report: Field Description Time Time in one-hour blocks if you chose Hourly or one-day blocks if you

chose weekly or monthly. The results show the utilization for each hour or day for the entire period shown in the “from” and “to” dates.

Percentage Gateway, route group, route list, or route pattern utilization percentage. This field gives the cumulative utilization percentage of the gateways or route groups or route lists or route patterns to the total number of calls that all the gateways put together can support at any one time.

6.2.1.1 Individual Trunk Line Counters Vendor Response Requirement Confirm that individual trunk line counters measure and report: Number of call attempts; Number of blocked trunk lines; Traffic intensity (Erlangs).

Cisco Response: Cisco complies. Refer to 6.2.1 above. 6.2.1.2 Outgoing Trunk Route Counters Vendor Response Requirement Confirm that outgoing trunk route counters measure and report: Number of outgoing attempts; Number of successful calls overflowing to another route; Number of lost calls due to blocking; Number of blocked trunks in measurement; Traffic intensity (Erlangs).

Cisco Response: Cisco complies. Refer to 6.2.1 above. 6.2.1.3 Incoming Trunk Route Counters Vendor Response Requirement Confirm that incoming trunk route counters measure and report: Number of incoming call attempts; Number of trunks in the measurement; Number of blocked trunks in the measurement; Traffic intensity (Erlangs).

Cisco Response: Cisco complies. Refer to 6.2.1 above. 6.2.1.4 Both Way Trunk Route Counters



Vendor Response Requirement Confirm that both way trunk route counters measure and report: Number of incoming call attempts; Number of trunks in the measurement; Number of blocked trunks in the measurement; Traffic intensity (Erlangs).

Cisco Response: Cisco complies. Refer to 6.2.1 above. 6.2.2 Attendant Consoles Attendant counters should measure all attendants in the system, or individual attendant positions. Record measurements include: number of answered calls; number of calls initiated by attendant; accumulated handling time for all calls; accumulated handling time for recalls; accumulated handling time for calls initiated by attendant; accumulated total delay time for recalls; number of answered recalls; number of abandoned attendant recalls; accumulated waiting time for abandoned calls to an attendant; accumulated waiting time for abandoned recalls, and accumulated response time for all types of calls. Vendor Response Requirement Confirm that attendant counters measure and provide reports for each of the listed parameters. Identify attendant parameters which are not measured.

Cisco Response: Cisco complies. See Response for 6.2.3 for Attendant Station level information by the Traffic Summary by Extension Report.

6.2.3 Stations Station counters should measure individual stations or station group traffic statistics, including: number calls; number of stations in measurement; number of blocked stations in measurement; traffic rating (Erlangs). Vendor Response Requirement Confirm that station counters measure and provide reports for each of the listed parameters. Identify station parameters which are not measured.

Cisco Response: Cisco complies. The CDR Analysis and Reporting Tool includes several reports that meet these requirements. These reports are the 1) Traffic Summary by Extension, 2) Traffic Summary System Overview. Traffic Summary by Extension Report: Field Description Time Indicates the cumulative hours of the day(s), the days of the week, or the

days of the month for the selected date range.

No of Calls Displays the percentage of calls for each gateway for the hours of the day, the days of the week, or the days of the month for the selected date range.

Internal Intracluster calls that originated in the Cisco Unified CallManager network and ended in the same Cisco Unified CallManager network (no gateways are



used).

Local Local calls that are routed through the public switched telephone network (PSTN) to numbers without an area code or which include one of the local area codes.

Long Distance Long-distance calls originating in the Cisco Unified CallManager network going out through the PSTN.

International International calls originating in the Cisco Unified CallManager network going out through the PSTN.

On Net Outgoing, intercluster calls that originate on one Cisco Unified CallManager cluster and terminate on a different cluster.

Incoming Inbound calls that originated outside the Cisco Unified CallManager network, entered through a gateway, and went into the Cisco Unified CallManager network.

Tandem Inbound calls that originated outside the Cisco Unified CallManager network, entered the Cisco Unified CallManager network through a gateway, and were transferred outbound from the Cisco Unified CallManager network through a gateway.

Others All other outgoing calls, such as toll-free numbers or emergency calls such as 911.

Total The total number of calls for each hour or day.

Traffic Summary System Overview Report: Field Description Top 5 Users based on Charge

Details the 5 users who have incurred the highest charges for calls that occurred during the specified date range.

Top 5 Destinations based on Charge

Details the 5 called numbers that have incurred the highest charges for calls during the specified date range.

Top 5 Calls based on Charge

Details the 5 calls that have incurred the highest charges for calls during the specified date range.

Top 5 User based on Duration

Details the 5 users who have spent the most time on calls during the specified date range

Top 5 Destinations based on Duration

Details the 5 called numbers that have been engaged in calls for the longest time during the specified date range.

Top 5 Calls based on

Details the 5 longest calls for the date range specified.



Duration

Traffic Summary Report - Hour of Day

Shows the volume of calls during the specified date range based on each hour of the day.

Traffic Summary Report - Day of Week

Shows the volume of calls during the specified date range based on each day of the week.

Traffic Summary Report - Day of Month

Shows the volume of calls during the specified date range based on each day of the month.

Quality of Service Report - Summary

Shows the number of calls that fell within each voice-quality category during the specified date range.

Gateway Summary Report

Shows the summary of the call classification for each gateway along with the QoS, the number of calls, and the duration for each classification for the gateway during the specified date range.

6.2.4 Traffic distribution When applicable, traffic distribution across the internal switching network should be measured for each local TDM bus, traffic over each highway bus, and traffic across the center stage switch by each switch network interface link. Vendor Response Requirement Confirm that traffic distribution is measured and reported for each switch network element listed. Identify what is not measured and reported.

Cisco Response: Not applicable. 6.2.5 Busy hour traffic analysis Busy hour traffic analysis measurements for trunks, stations, and the internal switch network should be performed and reported for any one hour interval for any time of the day. Vendor Response Requirement Confirm busy hour traffic measurements for trunks, stations, and the internal switch network for any one hour interval for any time of the day.

Cisco Response: Cisco complies. 6.2.6 Erlang Ratings



Erlang rating should be calculated and reported for individual trunk lines, each trunk group, and all trunk groups. CCS ratings should be calculated for individual stations or groups of stations. Vendor Response Requirement Confirm Erlang and CCS rating calculations and reporting for each listed item.

Cisco Response: Cisco complies. 6.2.7 Processor Occupancy System call processing performance is measured in terms of Busy Hour Calls (Attempts and Completions). The percent of maximum call processing capacity should be reported for programmed time intervals. Threshold reports should also be generated to monitor system load factors. Vendor Response Requirement Confirm measurement and reporting of processor occupancy and threshold levels

Cisco Response: Cisco complies. The collection and display of any system and device statistics concerning the current operation of the Cisco Unified CallManager system allows for a full understanding the state of the system without studying the operation of each of its components. The Real-Time Monitoring tool provides a variety of system variables in real time. It retrieves information that is contained in the database for each service, and it displays those statistics in meaningful configurations as defined by the administrator. The Real-Time Monitoring tool can also be customized to track additional objects and counters, configure alarms and traps, collect and analyze trace files and logs, generate reports and much more. Where the RTM tool provides real-time information, the CDR Analysis and Reporting tool provides the ability to search through historical data and generate reports about system call capacities and BHCA/BHCC. For example, where RTMT would allow you to see in real time how many calls are active, CAR would allow you to run reports on how many calls were attempted/connected last week.

6.2.8 Threshold Alarms For a variety of system hardware devices it should be possible to define a congestion threshold value, and measure generated alarms. Alarms are recorded in an Alarm Record Log. The types of devices that can be tracked include: tone receivers; DTMF senders and receivers; conference bridges; trunk routes; modem groups. Vendor Response Requirement Confirm recording and reporting of alarms for each listed item.

Cisco Response: Cisco complies. Administrators use Alarms to obtain the runtime status and state of the Cisco Unified CallManager system and to take corrective action to fix detected problems. Alarm definitions and levels can be customized and the alarms can be sent to multiple destinations: trace files, SNMP, Syslog and they are recorded in the RTMT alarm event viewer. Alarm



event levels include emergency, alert, critical, error, warning, notice, informational, and debug.

6.2.9 Feature Usage Feature usage counters for selected station features, e.g., call forward, call transfer, add-on conference, and attendant system features, e.g., recall, break-in, should be measured and reported for programmed intervals. Vendor Response Requirement Confirm recording and reporting of feature usage counters for both station and attendant operations.

Cisco Response: Individual feature usage is not specifically tracked. However, the call detail records would show the usage of certain features such as add-on conference, transfer, etc. and there is a CAR report for measuring the usage for Cisco Unified IP Phone Services which are subscribed by administrators and users for access.

Cisco Unified IP Phone Services Report Fields

Field Description

Cisco Unified IP Phone Services

The name of the selected service.

Number of Subscribers

The total number of subscribers for a given service.

% Subscription The percentage of users who have subscribed to a given service, out of the total number of subscriptions for all services.

6.2.10 VoIP Monitoring The management system should collect and store data to track usage and performance data of IP gateway devices, Unified IP Phones, and VoIP intercom/trunk calls. VoIP information reports may include: tracking of IP gateway devices and calls that pass through each gateway; gateway congestion; assignment of services or routes to gateways; tracking of phone numbers dialed or originating off-site numbers; and IP gateway addresses. Vendor Response Requirement Briefly describe all VoIP monitoring information records and reports that are available. Specify if VoIP QoS parameters such as jitter, call delay/latency, and packet loss are tracked and reported, and if a system administrator can monitor VoIP calls in real-time for QoS observing? Indicate if any third party equipment is being proposed as part of your solution.



Cisco Response: Cisco complies. Unified CallManager includes several standard reports and tools. Refer to section 6.2.1 for information tracked on the Unified IP Phones. IP Gateways used for connectivity to the WAN or PSTN are tracked in the various Gateway reports discussed in section 6.2.3. In addition, as an option VoiceCon could utilize Cisco Unified Operations Manager (IPCOM), Version 1.0, and CiscoWorks IP Communications Service Monitor (IPCSM), Version 1.0 to provide additional management and troubleshooting utilities. These are not included in the price of the proposal. Cisco Unified Operations Manager 1.1 is described in detail in the section below: Cisco Unified Operations Manager 1.1: Cisco® Unified Operations Manager 1.1 is part of the Cisco Unified Communications Management Suite. It provides a real-time service-level view of the entire unified communications solution and presents the current operational status of each element in the Cisco Unified Communications solution, including the underlying transport infrastructure. The application remotely polls and collects data from the various devices and provides diagnostic capabilities for faster trouble isolation and resolution. It continuously monitors the different elements such as Cisco Unified CallManager, Cisco Unified CallManager Express, Cisco Unity® software, Cisco Unity Express, Cisco Unity Connection, Cisco Unified Contact Center, Cisco Unified Contact Center Express, and Cisco Unified MeetingPlace® Express, as well as Cisco gateways, routers, switches, and IP phones. Cisco Unified Operations Manager does not deploy any agent software on the devices being monitored and thus is completely non-disruptive to system operations.

PRODUCT OVERVIEW Cisco Unified Operations Manager is part of the Cisco Unified Communications Management Suite, which provides a comprehensive and efficient solution for network management, provisioning, and monitoring of Cisco Unified Communications deployments while:



• Lowering management costs through intuitive, easy-to-use products

• Increasing productivity through contextual diagnostic tools that accelerate trouble isolation and troubleshooting

• Maximizing network reliability with network-wide operational tests and voice-quality monitoring and diagnostic tests

Cisco Unified Operations Manager 1.1 (Operations Manager 1.1) monitors and evaluates the current status of all the components of the Cisco Unified Communications solution including the underlying transport infrastructure. It remotely polls and collects data from the various devices in the deployment. It does not deploy any agent software on the devices being monitored and thus is completely non-disruptive to system operations. Operations Manager 1.1:

• Presents the current operational status of the Cisco Unified Communications solution and provides visibility through real-time service-level views of the entire solution.

• Increases the productivity of network managers by providing contextual diagnostic tools to accelerate trouble isolation and troubleshooting

-Through diagnostic tests, performance, and connectivity details about different elements of the unified communications solution -Using synthetic tests that replicate end-user activity and verify gateway availability as well as other configuration aspects of the unified communications infrastructure -Through IP Service Level Agent (SLA)-based diagnostic tests that can measure the performance of WAN links and measure node-to-node service quality -By providing actionable information in notification messages through context-sensitive links to more detailed information about service outages -By context-sensitive links to CiscoWorks products and Cisco Systems® management systems (when those are deployed), to provide the user with the broad and deep array of diagnostics capabilities

• Presents service-quality alerts by using the information available through Cisco Unified Service Monitor 1.1 when the latter is deployed. It displays mean opinion scores (MOS) associated with service quality between pairs of endpoints (the endpoints can be IP phones, messaging systems, conferencing systems, or voice gateways) at specified times involved in the monitored call segment and other associated details about the service-quality problem. When Cisco Unified Service Monitor reports a MOS threshold violation, Operations Manager 1.1 can further perform a probable path trace between the two endpoints and can report on any outages or impairments associated with intermediate nodes in the path.

• Provides current information about connectivity-related and registration-related outages affecting IP phones (both Session Initiation Protocol [SIP] and Skinny Client Control Protocol [SCCP]-based phones) in the network and provides additional contextual



information to determine the location and identification of the IP phones.

• Enables tracking of devices and IP phone inventory, tracks IP phone status changes, and creates a variety of reports that document move, add, and change operations associated with IP phones in the network.

• Provides flexible north-bound interfaces using Simple Network Management Protocol (SNMP) traps, syslog messages, and E-Mails that let Operations Manager 1.1 report the status of the network being monitored to a higher-level entity (typically a manager of managers).

• Operations Manager continues its industry-leading support for the Cisco Unified Communications family of products by adding support for newly released Cisco products such as Cisco Unified CallManager 5.0/4.2, Cisco Unity Connection 1.1, Cisco Unified MeetingPlace Express, Cisco SIP Proxy Server, and Cisco Unified Presence Server. Additionally, Operations Manager can discover and continuously monitor SIP phones in the unified communications deployment. In addition to discovering these SIP phones, Operations Manager can also support diagnostic tests to SIP phones and present an inventory of all the SIP phones in the deployment.

APPLICATIONS

Small and Medium-Sized Enterprises For small and medium-sized deployments (generally less than 1000 phones), the software component of Cisco Unified Service Monitor 1.1 (Service Monitor 1.1) can co-reside with Operations Manager 1.1 on a single platform. A single installation process installs all the necessary components. Operations Manager can monitor the deployment out of the box without the need to configure any rules or settings. Operations Manager 1.1 provides real-time notifications using SNMP traps, syslog notifications, and E-Mail that report the status of the network being monitored to a higher-level entity. Figure 1 shows the deployment model for small and medium-sized enterprises.

Deployment Model for Small and Medium-Sized Enterprises



CUCM-Cisco Unified CallManager, CUOM-Cisco Unified Operations Manager, CUSM-Cisco Unified Service Monitor, SRST-Survivable Remote Site Telephony, PSTN-Public Switched Telephone Network

Large Enterprises For large enterprise deployments (generally more than 1000 phones), it is recommended that Operations Manager 1.1 and the software component of Service Monitor 1.1 be deployed on separate platforms. Operations Manager 1.1 can be deployed centrally or in a distributed manner to scale to different deployment sizes. Each instance of Operations Manager 1.1 can manage multi-site and multi-cluster IP Communications environments. Operations Manager 1.1 can monitor the deployment out of the box without the need to configure any rules or settings. This software can also provide real-time notifications using SNMP traps, syslog notifications, and E-Mail that report the status of the network being monitored to a higher-level entity. Operations Manager 1.1 can also share device credential information with other CiscoWorks products if they happen to be deployed in the enterprise, providing better coordination for troubleshooting and resulting in reduced administrative overhead for network managers. Figure 2 shows the deployment model for large enterprises.

Deployment Model for Large Enterprises

CUCM-Cisco Unified CallManager, CUOM-Cisco Unified Operations Manager, CUSM-Cisco Unified Service Monitor, CUCME-Cisco Unified CallManager Express, CUE-Cisco Unity Express, SRST-Survivable Remote Site Telephony, PSTN-Public Switched Telephone Network


Service-Level View



The service-level view in Operations Manager 1.1 allows network managers to visualize their entire Cisco Unified Communications deployment. The service-level view is a real-time auto-refresh display that provides status information about all the unified communications clusters and the elements of the clusters in the deployment. Drill down views show the operational status of each element and its inter-relationships with other elements of the solution. This display serves as the central point to initiate a variety of actions that are available in Operations Manager 1.1. A context-sensitive right-click menu is provided through which network managers can get detailed status as well as historical information about the alerts on each of the elements. It is also possible to select each of the devices and initiate a variety of diagnostic tests, get access to graphical performance-monitoring and capacity-monitoring information, or get IP connectivity details for a selected device by launching a neighbor topology view that shows Layer 2 physical connectivity from the selected device. Operations Manager 1.1 also makes available a set of context-sensitive tools outside the application that can aid in further troubleshooting or diagnostics. Figure 3 shows the service-level view and its details for a multi-cluster unified communications deployment.

Service-Level View and Details for a Typical Multi-Cluster Deployment

Real-Time Alerts



Operations Manager 1.1 comes with built-in intelligence that can understand the role of every device in a Cisco Unified Communications deployment, and it monitors those devices for any kind of faults or outages. There is no need to write any rules to start monitoring; all the rules are built into the product. It also comes with factory-defined thresholds (which can be further tuned by network administrators) and an analysis engine that can detect the violation of any of these thresholds and immediately alert network managers through multiple means. These alerts are presented to the user through the Alerts and Events Display, which refreshes periodically to present the most up-to-date status of the monitored devices. A separate display called the Phone Status Display provides instant access to IP phone outage information. Two types of outages are monitored: signaling-related outages and IP connectivity-related outages. It is also possible to get information about an IP phone's switch and port, allowing administrators to troubleshoot problems that may have wider scope (at the switch level) than just the IP phone. Figure 4 shows real-time alerts in the Alerts and Events Display.

Real-Time Alerts as Displayed in the Alerts and Events Display

Diagnostic Tests Operations Manager 1.1 comes with a rich set of diagnostic tests that can be used to aid in trouble isolation and resolution. There are primarily three types of tests: synthetic tests, phone status tests, and node-to-node IP SLA tests. The synthetic tests serve to replicate user activity (getting dial tone, making phone calls, leaving voice mail, and creating or joining conference calls). These tests can verify the functional availability of the supporting infrastructure and validate different configuration aspects such as route patterns, route lists, inter-cluster trunks, and gateway dial peers. Such synthetic tests can be performed using both the SIP and the SCCP signaling protocols. The phone status tests can be used to determine the current operational status of the IP phones in terms of signaling (SIP and SCCP) and IP connectivity. The node-to-node tests use the services of the Cisco IP Service Level Agent (IP SLA, formerly known as Service Assurance Agent [SAA]) in Cisco routers to simulate traffic in the network and then determine



network characteristics such as reachability status, response time, latency, jitter, packet loss, and network quality. Each of these tests can be run in a continuous monitoring mode as well as scheduled or on-demand modes. For example, the quality of a WAN link between different sites can be tested using the node-to-node tests and when the quality of the link drops below a certain threshold, an alert can be generated. Similarly, end-user tests can be set up that place test calls between different sites in an automated routine. The results of these diagnostics tests are presented through a variety of reports, and alerts can be triggered based on key thresholds being exceeded.

Service-Quality Reporting Operations Manager 1.1 can use the information provided by Service Monitor 1.1 to present service-quality (such as quality-of-voice) alerts on a real-time basis. The service-quality alerts are associated with IP phones or unified communications devices that are currently monitored by Operations Manager 1.1 and present that information in the Service Quality Alerts Display. Details about IP connectivity of the IP phones and devices are available to enable further troubleshooting. It is also possible to initiate a probable path trace between the endpoints that helps network managers identify any potential problems in intermediate nodes that could influence service quality. Figure 5 shows service quality alerts.

Service Quality Alerts

Reports Operations Manager 1.1 provides an extensive set of reports that help network managers maintain information about their Cisco Unified Communications deployment. The historical alert, event, and service-quality reports maintain information about all the alerts and events reported by Operations Manager 1.1 for up to 30 days. This enables network managers to document any past outage and have access to it for long-term trending purposes. The IP phone inventory reports give network managers instant access to status information about every IP phone deployed in the network.



Extensive information on signaling details and IP connectivity details is maintained and reported. These reports also track changes in phone status and thus serve to document move, add, and change operations on these IP phones. Such reports are available for both SIP- and SCCP-based IP phones. The customizable reports let network managers choose what type of information they want and create a daily report that is available by E-Mail or the Operations Manager 1.1 GUI.

PRODUCT ARCHITECTURE Operations Manager 1.1 is a Web-based application. It uses open standards-based access to gather operating status information from Cisco Unified Communications applications and Cisco IOS® Software to provide the information required to manage increasingly complex unified communications environments. Operations Manager 1.1 does not deploy any agent software on any platform it monitors. It uses open interfaces such as SNMP and HTTP (AVVID XML Layer-AXL) to remotely (and periodically) poll the devices being monitored and thus collect status information. It also performs several diagnostic tests (based on SCCP, SIP, and Cisco IP SLA) and uses the results to determine the operational status of the monitored devices. The user interface is browser-based to enable remote login from anywhere in the network and allow instant access to real-time information on the current status of the devices. Different levels of user access can be set up locally or in conjunction with Cisco Secure Access Control Server, which controls access to information in Operations Manager 1.1. Figure 6 shows the product architecture of Operations Manager 1.1.

Cisco Unified Operations Manager 1.1 Architecture

PRODUCT SPECIFICATIONS The table below shows Cisco Unified Operations Manager 1.1 product specifications:



Description Specification

Product compatibility

Cisco Unified Communications deployments consisting of Cisco Unified CallManager (including CallManager 5.0/4.2), Cisco Unity, Cisco Unity Connection, Cisco Unified Contact Center, Cisco Unified MeetingPlace Express, Cisco Unified CallManager Express, Cisco Unity Express, Cisco Unified Contact Center Express, Cisco Conference Connection, Cisco Personal Assistant, Cisco Emergency Responder, routers, gateways, switches, and IP phones

Software compatibility

· Windows 2003 Server · The user interface can be accessed using Microsoft Internet Explorer 6.0 on Windows

2003 and Windows XP platforms.

Protocols Uses SNMP, SCCP, and HTTP (Cisco AVVID XML layer-based) to monitor the unified communications deployment

Features and functions

Automatic device and phone discovery, service-level view, real-time alerts, diagnostic tests, service-quality alerting, endpoint status and endpoint status change reports, north-bound interfaces, performance and utilization monitoring, historical alerts, event and service-quality reports, context-sensitive launch of CiscoWorks products

SYSTEM CAPACITY (per Operations Manager Instance)

System Parameter Capacity

Monitored phones 1000 10,000 30,000

Monitored devices 300 1000 2000

Monitored Cisco Unified CallManager clusters 10 15 30

Monitored Cisco Unified CallManager Express routers 100 250 500

Monitored Survivable Remote Site Telephony (SRST) routers 10 100 500

Concurrent synthetic tests 25 100 250

Concurrent node-to-node (Cisco IP SLA) tests 25 100 250

Concurrent client (browser) logons 5 5 5

For unified communications deployments of more than 30,000 phones, multiple Operations Manager 1.1 servers can be deployed. These servers can share device and credential information between them and administrators can perform centralized device and credential management.



By integrating the Operations Manager with a Cisco Secure Access Control Server, administrators can centrally control user access. Each of these Operations Manager servers will roll up the status of the network being monitored to a higher-level entity (typically a manager of managers) through SNMP traps and syslog notifications.

Features • Automatic device and phone discovery

• Service-level view of the complete Cisco Unified Communications deployment with current status information on all monitored elements

• Real-time alerting on all the monitored unified communications devices

• Diagnostic tests such as end-to-end synthetic tests, node-to-node Cisco IP SLA tests, and phone status tests

• Service-quality alerting based on information from Cisco Unified Service Monitor 1.1 or Cisco Unified Service Monitor 1.0

• Endpoint status and endpoint status change reports (for both SIP- and SCCP-based IP phones)

• Northbound interfaces using SNMP traps, syslogs, and E-Mail notifications with context-sensitive links to more detailed information

• Performance and usage monitoring of various Cisco Unified Communications devices

• Historical alert, event, and service-quality reports

• Context-sensitive launch of CiscoWorks products if they are deployed

6.3 Optional Reports Directory records may include each subscriber’s name along with a variety of phone numbers such as primary, published, listed, emergency, and alternate, as well as authorization code information, job title, employee number, current employment status and SSN.

Inventory records and management is used to administer any kind of inventory product part, including: PBX common equipment (cabinets, carriers, circuit cards); voice terminals and module options; jacks, and button maps. The reports allow administrators to accurately re-charge items. Inventory can be tracked by data such as user, system (PBX or other networks), jack, serial number, asset tags, trouble calls, recurring and non-recurring costs, and general ledger codes. The inventory management system may also include records containing the following data: purchase date, purchase order number, depreciation, lease dates, manufacturer and warranty information.

Cabling records keep track of all cable, wire pairs, distribution frames, wiring closets and all connections (including circuits) down to both the position and the pair level. Cable records include starting and ending locations, description, type and function. Individual cable lengths are maintained and automatically added, as is the decibel loss, for the entire path. Information



can also be provided on the status of all cable runs, as well as the number of pairs it contains, the status of the pairs, and the type of service it provides. Vendor Response Requirement Identify and briefly describe your proposed management system’s Directory, Inventory, and Cabling reports, if available.

Cisco Response: Cisco Directory Records may be extracted by utilizing the Bulk Administration Tool (BAT) that will allow an administrator with permissions to Export all of the Unified CallManager User/Phone data to a CSV File format.

Several Cisco Technology Partners offer operations and management applications that include cable management. Cisco does not have a product in this category.

Inventory management is part of the Cisco Unified Operations Manager optional package described above in Section 6.2.10.

6.4.0 Call Detail Recording

Call Detail Record (CDR) data should be compiled for all successful incoming and outgoing trunk calls. Call record fields typically include the following:

• Date • Time • Call Duration • Condition Code (categorizes information represented in the call record) • Trunk Access Codes • Dialed Number • Calling Number • Account Code • Authorization Code • Facility Restriction Level for Private Network Calls • Transit Network Selection Code (ISDN access code to route calls to a

specific inter-exchange carrier) • ISDN Bearer Capability Class • Call Bandwidth • Operator System Access (ISDN access code to route calls to a specific

network operator) • Time in Queue • Incoming Trunk ID • Incoming Ring Interval Duration • Outgoing Trunk ID

Vendor Response Requirement VoiceCon will purchase its own third party call accounting and billing system. Identify all available CDR reports that can be generated for any or the entire call record field data listed above.



Cisco Response: Cisco continues to expand the Cisco Technology Partner Program. Currently, numerous vendors have updated their billing and accounting applications to be compatible with the Cisco Unified CallManager CDR format. The table below shows the current list of partners. There is also a searchable database at the following url for information about partners in other application areas: http://forums.cisco.com/eforum/servlet/IPCApps?page=main



In addition, if a customer has a preferred vendor not on this list, the CDR and CMR record formats can be provided so that the vendor can modify their application to use Cisco Unified CallManager SQL data. Call Detail Reporting (CDR) is a standard feature of Unified CallManager. Cisco Unified CallManager can create Call Detail Records (CDR’s) and Call Management Records (CMR’s). These are intended to help administrators and others responsible for billing, record keeping, and problem mitigation to have available a record of all calls that have been originated by or terminated by end users of the Cisco Unified CallManager. CDR and CMR options are configured in Cisco Unified CallManager Administration application. You can configure the following CDR and CMR options:

• Whether or not the CDRs are created • Whether or not the CMRs are created • How often to write CDRs to disk

For every call that is initiated or terminated by a user of the Unified CallManager system, there will be multiple records written for each call. Two are required, and they are the StartCall record, and the EndCall record. There may be several other records written for a given call depending on what happened to the call. CallStart record is written at the start of each call and a CallEnd record is written when the call is terminated. If call diagnostics are enabled, CMR records are normally written for each call when a call involves an Unified IP Phone as an endpoint. If both endpoints are Unified IP Phones, then there will be two CMR records written, one for each phone. CMR data will not be generated unless both CDR and Call Diagnostics are enabled, but CDR data may be generated and logged without CMR data. For many of the more complex types of calls that involve one or more supplementary services, several CDR’s will be generated. CDR data is produced by the Call Control within the Unified CallManager. The records are written when significant changes occur to a given call, such as starting a call, ending a call, transferring the call, redirecting the call, creating or joining a conference, etc. CDR data from multiple Unified CallManagers in a cluster is collected on one designated Unified CallManager server, rather than being collected separately in each of them.

Avotus Corporation - InteleControl MVP

Avotus Corporation - Avotus Professional

Avotus Corporation - Avotus ICM

Avotus Corporation - InteleControl for Windows v2.1 UNIX v6.3

ISI Telemanagement Solutions - Infortel Select

ISI Telemanagement Solutions - Infortel Select for Cisco Unified CallManager Express

Resource Software International - Shadow CMS with Winlink for Cisco Unified CallManager v1.0

Soft-ex Communications Ltd - Soft-ex Telephony Manager v3.0a

Soft-ex Communications Ltd - Soft-ex Telephony Manager (STM)

Soft-ex Communications Ltd - Soft-ex ICMS (Integrated Communication Management Service), Version 3.1b

STONEVOICE SRL - Stonevoice BILLY, version 2.1

VERAMARK TECHNOLOGIES - Veramark eCAS Version 2.4

VERAMARK TECHNOLOGIES - Veramark eCAS Version 2.4



The raw data for Call Detail Records (CDRs) and Call Management Records (CMRs) are saved in ASCII text files and periodically uploaded via sFTP to a network share for permanent storage and post processing. This same raw data is also imported into the Unified CallManager SQL database for running queries and generating reports using the CAR tool. There is no real limit to the number of records that can be collected, though it is recommended to purge the database periodically and the CAR tool provides for configurable thresholds and purging schedules.

6.5.0 Maintenance System maintenance operations should, at minimum, support the following: Monitoring of processor status; Monitoring and testing of all port and service circuit packs; Monitoring and control of power units, fans, and environmental sensors; Monitoring of peripherals (voice terminals and trunk circuits); Initiate emergency transfer and control to backup systems; Originate alarm information and activate alarms. Vendor Response Requirement Confirm support of each listed maintenance monitoring activity. Identify any activity not supported.

Cisco Response: Cisco complies. 6.5.1 Alarm Conditions There are usually several types of communications system alarm conditions: Major, Minor, and Warning. Vendor Response Requirement Briefly describe how your management system defines a Major, Minor, and Warning alarm.

Cisco Response: Cisco complies. Alarm definitions and levels can be customized and the alarms can be sent to multiple destinations: trace files, SNMP, Syslog and they are recorded in the RTMT alarm event viewer. Alarm event levels include emergency, alert, critical, error, warning, notice, informational, and debug.

6.5.2 Maintenance Reports Vendor Response Requirement Identify any and all available maintenance alarm reports provided by your management system.

Cisco Response: Cisco complies. You can use the Real-Time Monitoring tool to view real-time and historical alarm information, collect log and trace files, etc.

6.5.3 Remote Maintenance Vendor Response Requirement



Briefly describe the available options used to support remote maintenance operations for both customer access and for an outside maintenance service provider. Specify how the system alerts a remote service center when an alarm condition occurs, the trunk circuit requirements for alert transmissions, and security measures to prevent unauthorized access.

Cisco Response: Remote Serviceability. Cisco Service Engineers (CSE) use the remote serviceability tools to supplement the management and administration of your Cisco Unified CallManager system. Using these tools, CSEs gather system and debug information when remote troubleshooting or diagnostic help is needed. With customer permission, technical support engineers log on to a Cisco Unified CallManager server that allows them to perform any function that could be done from a local logon session. Remote serviceability supports numerous applications in the multihost and multiplatform Cisco IP Telephony Solutions environment. The tools can process and report on a vast collection of local or remote Cisco Unified CallManager configuration data and system information. Cisco Unified CallManager supports the following capabilities for remote serviceability:

• Cisco Secure Telnet—Allows CSEs to log on to customer remote site to troubleshoot Cisco Unified CallManager system.

• Command Line Interface—Allows CSEs to display Cisco Unified CallManager system statistics and troubleshoot the Unified CallManager platform on customer network.

• Real-Time Monitoring tool—Allows CSEs to collect log files, view real-time statistics

• Message Translator for ISDN Trace—Allows CSEs to use Q931 Message Translator to debug ISDN Layer 3 protocol messages.

• CiscoWorks2000 Network Management System—Provides remote network management for a Cisco Unified CallManager cluster.

• Path Analysis Interface—Traces connectivity between two specified points on a network and analyzes both physical and logical paths (Layer 2 and Layer 3) taken by packets flowing between those points.

• System Log Management—Provides a centralized system logging service for Cisco IP Telephony Solutions.

• SNMP Instrumentation—Enables administrators to remotely manage network performance, find and solve network problems, and plan for network growth.

• Cisco Discovery Protocol Support—Enables discovery of Cisco Unified CallManager servers and management of those servers by CiscoWorks2000.

Cisco Secure Telnet Cisco Secure Telnet offers Cisco Service Engineers transparent firewall access to Cisco Unified CallManager servers on the customer’s site. Cisco Secure Telnet works by enabling a Telnet client inside the Cisco Systems firewall to connect to a Telnet daemon behind the customer’s firewall. This secure connection allows remote monitoring and maintenance of the Cisco Unified CallManager servers without requiring firewall modifications.

6.6.0 Provisioning



All services should be provisioned in one step. Services should include station configuration, voice mailbox configuration, E-911 location, billing attributes, directory attributes, and mobile Email attributes (Blackberry) and the configuration of other end user applications. For example, if your solution includes a zone paging application, the ability to assign a station to a zone and change the zone membership as a whole must be accessible through the configuration (provisioning) interface. Templates must be supported to organize different settings across different systems according to organizational need. At a minimum, the voice station configuration and the associated voice mailbox must be provisioned in one step through one interface. Your proposed provisioning application or interface must create a complete audit trail and must allow groups of changes to be scheduled for a future time. Further, the solution must support mass create, delete and modify functions to support bulk operations. Vendor Response Requirement Describe the provisioning workflow you recommend showing how each of your proposed solution components is utilized. List any functions above which are not available. List any systems or devices which are not now part of your provisioning interface and provide a roadmap statement of how you will treat this situation going forward.

Cisco Response: The Cisco Voice Provisioning Tool (VPT) has been included in this proposal. Cisco VPT is a unified set of provisioning interfaces and services that make the initial setup and ongoing administration of Cisco Unified CallManager and Cisco Unity faster, easier, and more efficient. Cisco VPT provides a system-level approach to telephony management by combining the most common user attributes from multiple Cisco Unified CallManager and Cisco Unity servers. This simplified interface maximizes productivity for administrators by enabling them to manage common daily administrative tasks such as Moves, Adds and Changes from a single console.

By providing a single point of entry for administrators, Cisco VPT creates a simple, workflow-based environment that makes managing IP Communications deployments more intuitive. Cisco VPT intelligently tracks Cisco telephony and messaging user data across multiple servers, sites and clusters on the network—even ones that are running different software versions. Cisco VPT allows extensive use of templates, cutting down the number of fields that need to be manually entered and dramatically reducing the number of administrative errors. For network troubleshooting, system managers can use the single unified interface of VPT to rapidly and easily view all of the data associated with any subscriber or phone in the network and get right to the heart of the problem.



For added security, Cisco VPT features roles-based access, so managers can grant different permission levels to system administrators. Cisco VPT also provides audit logs to track which administrators made changes when, and what changes were made to each record.

Voice Provisioning Tool Features Supported

Subscriber Management Features

• Search for, view and sort subscribers based on name, username, extension and other key parameters

• Add, delete and modify subscriber records for both Cisco Unity and Cisco Unified CallManager

• Associate a subscriber with an Unified IP Phone or create the phone from within the subscriber page

• Multi-Modify: Select a group a subscriber records and apply a change to all subscribers in the group with a single action

• Templates: Use subscriber templates to rapidly and easily create subscribers with similar features

Cisco Unified IP Phone Management Features

• Search for, view and sort Cisco Unified IP Phones based on various key parameters

• Add, delete and modify Cisco Unified IP Phone data

• Add, delete and modify lines on an existing Cisco Unified IP Phone

• View shared lines and modify shared line settings on a Cisco Unified IP Phone

• Subscribe or un-subscribe to Unified IP Phone services

• Restart or reset Cisco Unified IP Phones directly from the VPT interface

• Multi-Modify: Select a group a Cisco Unified IP Phone records and apply a change to all Unified IP Phones in the group with a single action

• Templates: Use Unified IP Phone templates to rapidly and easily create Unified IP Phones with similar features

Administrative Features

• Web browser-based interface: Use the lightweight, web browser interface to access the Cisco VPT server from your laptop virtually anywhere.

• Secure provisioning: Ensure that your subscriber provisioning data is secure using HTTPS, IP Sec and TLS to build an end-to-end secure system.

• Audit logs: Track the date, time, administrator who made the change, and the subscriber or phone record that was edited using extensive VPT audit logs.

• Bulk Administration: Use spreadsheet based comma separated value (CSV) files to import or export large numbers of subscribers or phone



records. Schedule the date and time you want these changes to take place to mange the impact on your IP Communications system.

• Role based access control: Define different levels of administrative privileges and assign administrators to these roles. Privileges can be configured by Unified CallManager or Unity server and by read or write capabilities.

• Supports up to 20 system administrators simultaneously making changes on the IP Communications system.

• System administrators may use native product user interfaces or Cisco VPT with no risk of data inconsistency.

• Centrally manage IP Communications systems with up to 10 Cisco Unified CallManager clusters and up to 10 separate Cisco Unity systems. These Cisco Unified CallManager clusters and Cisco Unity servers may be remotely located.

• Manage IP Communications systems with multiple different software versions of Cisco Unity and Cisco Unified CallManager. Simply download and install new software plug-ins for Cisco VPT to connect to and manage new versions of supported products.



7.0.0 Integrated Messaging System VoiceCon requires a HQ-based voice messaging system that must be fully integrated with the proposed IPTS network solution. VoiceCon also requires integration of the proposed voice messaging system with a Microsoft Exchange messaging system to provide “unified” messaging applications. The proposed voice messaging system solution must be centrally located at the VoiceCon HQ location, and be capable of supporting station users at all remote VoiceCon faciliities (RO and SBs). The voice mail system will also serve as an automated attendant position for select incoming trunk calls, and also as a secondary point of coverage as an automated attendant system for designated stations. All software and hardware necessary to interface with the existing telephone system will be provided under this bid. The sizing requirements are:

Five (5) automated attendant ports are included in the requirements. A Grade of Service level of P.01 is required. Vendor Response Requirement Briefly describe the proposed integrated messaging solution, and provide details about the voice mail system architecture and it’s interconnection to the voice communications system and Microsoft Exchange system. Include processing system platform information in the discussion. Verify that the system being bid can comply with each of the proceeding requirements.

Installed/Equipped Capacity Maximum Capacity Number of Users 2000 3,000 Number of Ports 64 96 Hours of Storage 1000 1200



Cisco Response: Cisco is proposing Cisco Unity Release 4.2 to meet the unified messaging and automated attendant requirement. Unity Voice Mail is a Windows 2000-based communications solution that delivers voice mail in a networked business environment. Unity Voice Mail gives users the ability to access and manage their voice mail from a touchtone telephone, and control voice mail options from an Internet Explorer interface.

Cisco Unity® Unified Messaging—an integral component of the Cisco IP Communications system—is a foundational element in bringing unified communication solutions to enterprise-scale organizations. It delivers powerful unified messaging (E-Mail, voice, and fax messages sent to one inbox) and intelligent voice messaging (full-featured voice mail providing advanced functions) to improve communications, boost productivity, and enhance customer service capabilities across your organization. Cisco Unity Unified Messaging and Voice Messaging increases productivity by providing anytime, anywhere access to all types of messages. It lowers costs by helping enable a flexible migration to IP Communications and taking advantage of existing voice, data, and groupware investments. Many Cisco Unity customers save as much as one hour per user per day in productivity gains, along with lowered operational expenses associated with converged voice and E-Mail directories and services.

Cisco Unity for Lotus Domino and Microsoft Exchange environments features:

• Delivers powerful unified messaging in a unified Lotus Domino environment.

• Delivers advanced voice mail and powerful unified messaging in a unified Microsoft Exchange environment.



• Natively supports Session Initiation Protocol (SIP) proxy servers, designated Unified IP Phones and clients, and SIP-enabled access gateways.

• Live Reply enables subscribers to immediately reply to messages from other subscribers by calling them back directly from the telephone user interface.

• Flex Stack allows subscribers to specify the order in which they want messages presented to them over the phone, whether by message type (voice, fax, E-Mail), urgency, or LIFO/FIFO (last in first out/first in first out).

• Unity Assistant (the Cisco Personal Communications Assistant’s browser-based personal administrator) allows IT staff to enable end users to manage more of their own accounts, saving time and decentralizing routine administration.

• Fault-tolerant system tools include robust security, file replication, event logging, and optional software RAID levels 0-5.

• International product offering fully localized versions in U.S. English, French, German, and Japanese. Full localizations feature system prompts, subscriber conversations, browser-based administration consoles, and product documentation in the customer’s language of choice.

• Localized telephone system prompts available in multiple languages, including four dialects of English (Australian, New Zealand, U.K. and U.S.), two dialects of Chinese (Mainland Mandarin and Taiwan Mandarin), Danish, Dutch, Italian, Korean, Norwegian, Brazilian Portuguese, two dialects of Spanish (Colombian and European), and Swedish.

Cisco Unity for Microsoft Exchange environments features:

• VPIM support provides digital interoperability with legacy voice mail systems, providing advanced interoperability features, faster message delivery, and lower cost with more efficient message transmission.

• Cisco Unity Bridge provides advanced message interchange with legacy Avaya/Octel voice mail systems—unlocking proprietary networking to deliver open standards-based IP migration.

• Unity Inbox—a message access console (supported on Internet Explorer 5.5 or higher)—provides a dedicated voice mail inbox to deliver unified messaging functionality to non-Exchange network environments (i.e. Novell GroupWise, etc.).

• Failover prevents service disruption if unified messaging server is down, delivering enhanced reliability and serviceability.

Cisco Unity Voice Mail integrates to Unified CallManager through a SCCP/TAPI connection. Unity Voice Mail is available in various session configurations supporting up to 96 ports and 4,000 users (with internal message storage) or 7,500 users (with external message storage) and up to 2,245 hours of message storage with G.711 or 17,964 hours with G.729 voice compression. For this RFP Cisco has quoted one 72-session configuration installed as a turnkey system on one Cisco MCS 7845-H2 server. This turnkey solution



comes completely assembled and tested with all the software installed. This configuration provides 96 ports installed capacity for growth. Unity Voice Mail - Features: The next several pages summarize the features available with Unity Voice Mail. Intelligent Voice Mail Intelligent Voice Mail reflects the latest research on how users interact with the voice mail system. With its VMUIF-compliant interface, stream-lined system conversation, and multiple menu options, Unity Voice Mail’s Intelligent Voice Mail allows internal subscribers and outside callers to reach employees, leave messages, receive information, and handle voice mail faster and more efficiently than ever before. What does a VMUIF-compliant interface mean to you? The Voice Mail User Interface Forum determines the industry-standard menus for controlling voice mail with touchtone keystrokes. It means that employees switching to Unity Voice Mail from other voice mail systems or working in mixed voice mail environments under Unity Voice Mail will already be familiar with the basic keystroke operations. Changing systems will be easy and new employees won’t have to spend hours of training time learning to use Unity Voice Mail. If you are an experienced voice mail user, Unity Voice Mail lets you make use of the system’s powerful features while simultaneously minimizing your interaction with the system itself. Automated Attendant The automated attendant functions give Unity Voice Mail the ability to act as an electronic receptionist, answering and routing incoming calls automatically. Callers hear an opening greeting that provides instructions, information, and options. They can then reach a subscriber directly by dialing the extension number at any time during the opening greeting. Callers who do not know the extension number of the party they are trying to reach can access a directory. By pressing keys corresponding to the letters of the person’s name, Unity Voice Mail can identify the subscriber and transfer the caller to the subscriber’s extension—either immediately or after giving the caller the subscriber’s extension number for future reference. Cisco Unity Voice Mail can answer several calls at once, place callers on hold inform them of their position in the holding queue, and periodically update this information. It can also screen calls announce the name of the caller, and wait for confirmation from the called party before putting the call through. In this way, Unity Voice Mail’s automated attendant functions can handle a high volume of routine incoming calls faster and more efficiently than a live operator. And the operator is left free to provide personalized customer service to those who need it most. Cisco Personal Communication Assistant (CPCA) Cisco Unity Voice Mail gives voice mail or unified messaging subscribers the ability to customize their personal settings from Internet Explorer 4.01 or higher using Cisco CPCA, a dynamic Web-browser interface based on Microsoft’s Active Server Page (ASP) technology. CPCA reduces the workload



for system administrators and gives subscribers additional flexibility to customize Unity Voice Mail to suit changing demands in their work environment. Subscribers can quickly and easily establish or change personal settings, such as their voice mail options, passwords, private distribution lists, and message delivery options. Users can also reset their TUI (Telephone User Interface) passwords themselves, significantly reducing the workload of the help desk for such tasks and thus reducing administrative costs. • Administer private distribution lists: Subscribers oversee their private

distribution lists, adding or deleting recipients as necessary. • Change directory listing status: Individual subscribers can control

whether or not they want to be listed in the directory. • Change password: Subscribers can change their own passwords. • Change transfer options Subscribers can turn call transfer on or off,

and change the extension or telephone number to which incoming calls are transferred.

• Record/Edit: From a PC or telephone, subscribers can record their directory name, and record and edit their personal greetings.

• Set conversation options: Subscribers can specify whether or not they want full or brief menus, or to hear the day and time stamp before or after a message, or if they want the voice mail system to greet them by name when they call in to retrieve their messages.

• Set message delivery options: Subscribers can establish rules governing message delivery options for the type and urgency of message or frequency of delivery

• Set Alternate Phone Numbers: Subscribers can add alternate telephone numbers that allow the caller to call from their Mobile or Home phone and be recognized by the system as a user.

7.1.0 Support for Open Standards Vendor Response Requirement Describe voice messaging system’s support for open standards. List the clients that can be used with your proposed solution. For proprietary clients, detail minimum hardware and software requirements

Cisco Response: The primary goal of open standards support is to ensure wide interoperability/integration within multi-vendor environments. Cisco Unity is an industry leader in supporting integrations with a broad range of industry voice mail system, TDM-based PBX's and even VoIP based systems. To do this, Cisco has implemented both AMIS and VPIM for voice message exchange with other voice mail systems, analog DTMF, PBX-Link Digital Set Emulation and SMDI as standard integration methods with TDM PBX's and SIP as the primary standard for call control/integration in a VoIP environment. Cisco Unity also supports interconnectivity to an Octel Network using the Cisco Unity Bridge. Cisco Unity can also work with E-Mail clients that support Simple Mail Transfer Protocol (SMTP)/ Multipurpose Internet Mail Extensions (MIME) and Internet Message Access Protocol 4 (IMAP4).



Cisco Unity supports SIP via a SIP proxy or by integrating with the Intel PBX IP Media Gateway (PIMG), which can support serial integration, analog in-band DTMF integration and digital feature set emulation. Cisco Unity supports many open standards and some vendor specific proprietary standards, such as Microsoft’s MAPI protocol and Lotus Notes API used for connectivity between Cisco Unity and Microsoft Exchange and Lotus Domino, respectively. Cisco Unity supports IMAP4 client access in a voice mail integrated messaging configuration, for E-Mail clients that support the IMAP4 protocol. When configured to service MS Exchange, Cisco Unity supports a client snap-in called ViewMail for Outlook which can be snapped into MS Outlook to provide for a rich unified messaging experience. When configured to service Lotus Domino Cisco Unity supports a Notes client snap-in called Domino Unified Communications Service client to provide for a rich unified messaging experience. Cisco Unity also supports web access to the Cisco Unity Inbox via its Cisco Personal Communications Assistant web application available on the Unity server. Through the Cisco Unity Inbox, clients can listen to and manage their voice messages through a web interface. The supported Cisco Unity client matrix is included to provide a list of third party clients supported:

Cisco Unity with

Exchange

Operating System on

Workstation

Cisco Unity ViewMail for

Microsoft Outlook on

Workstation

Messaging Client on

Workstation

Internet Browser on Workstation

4.0(5) Windows • XP • 2000 • NT 4.0 • ME • 98

4.1(1) 4.0(x) 3.1(x) 3.0(x)

Outlook • 2003 • 2002

(XP) • 2000 • 98

Internet Explorer

• 6.0 (plus Service Pack 1)

• 5.5 (plus Service Pack 2)

Cisco Unity with

Domino

Operating System on

Workstation

IBM Lotus Domino Unified Communications (DUC) for Cisco on Workstation

Messaging Client on

Workstation

Internet Browser on Workstation

4.0(5) Windows • XP • 2000 • ME • 98

1.2.3 Lotus Domino Web Access

• 7.0 Lotus iNotes

• 6.5.x • 6.0.x

Lotus Notes • 7.0 • 6.5.x

Internet Explorer • 6.0 (Plus

Service Pack 1)

• 5.5 (Plus Service Pack 2)



• 6.0.x 4.0(5) Windows

• XP • 2000 • ME • 98

1.2.2 Lotus iNotes 6.5x 6.0x 5.0.12 Lotus Notes 6.5x 6.0x 5.0.10 and later

Internet Explorer • 6.0 (plus

Service Pack 1)

• (plus Service Pack 2)

Please NOTE: Some caveats exist. In addition, more detail is provided at the following link: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_unity/cmptblty/clientmx.htm

7.1.1 Security Features Vendor Response Requirement Describe security features available with the voice messaging system to prevent abuse and unauthorized access.

Cisco Response: Unity Voice Mail ensures the safety and integrity of the unified communications environment by offering many advanced security features, such as secure private messaging, two-factor TUI authentication, standard account lockout functionality and securing the connection between Cisco Unity, Unified CallManager and Unified IP Phones.

• Hacker detect and lock accounts Unity Voice Mail monitors the number of attempts made to log on to an account via the telephone and can lock accounts if incorrect passwords are entered repeatedly. The system administrator can specify the number of invalid log-on attempts allowed and the number of minutes before the system resets the account lockout. Locked accounts can be unlocked manually by the system administrator or automatically by Unity Voice Mail after a specified number of minutes.

• Passwords The system administrator can define the number of characters for passwords and how often Unity Voice Mail will require a password to be changed.

• Trivial Password Protection Unity Voice Mail can be configured to check for trivial passwords and can restrict the use of them.

• RSA Secure ID Two factor security for TUI access Cisco Unity supports the RSA two factor authentication for TUI access. The RSA two-factor authentication allows users to use token fobs to enter unique passcodes each time they login to the Unity system via the telephone user interface (TUI). Unity supports the RSA Ace server and token fobs which must be obtained directly from RSA.

• Cisco Security Agent (CSA) Cisco Security Agents for Cisco Unity, the Cisco Unity Bridge, and Cisco Personal Assistant. These security agents protect the application and the operating system by blocking malicious attacks, such as buffer overflows,



Trojan horses, malformed packets, and malicious HTML requests. • Preventing toll fraud develop preventive measures, and best

practices to avoid toll fraud using restriction tables in Cisco Unity • Password and Account Policy Management Cisco Unity

provides guidance on managing accounts and their passwords. • Secure Private Messaging By using this feature, messages

marked private cannot be forwarded by phone. This includes any voice message that a Cisco Unity subscriber marked private. The voice message wav file is actually encrypted and only a Unity server can playback the message to the intended subscriber. If E-Mail is used to forward the voice message, the message cannot be played back on a computer or other messaging system. A decoy message is included as a part of the encrypted message.

• Securing the connection between Unity, Unified CallManager and Unified IP Phones Cisco Unity supports security features to Unified CallManager and Unified IP Phones that use TLS to provide signaling authentication, device authentication via the creation of the Cisco Certificate Trust List (CTL) file, Signaling encryption of all SCCP messages between Unity and Unified CallManager and Media encryption by using sRTP as defined in RFC 3711.

• SSL support for client web connections SSL can be used as a method of providing security for transmission of Cisco Unity data across the network through the use of public/private key encryption. SSL protects the security of Cisco Unity subscriber credentials when they are passed across the network. SSL also protects the security of all data entered in Cisco Unity web applications.

7.2.0 Voice Mail Features 7.2.1 Forwarding The system must provide access for forwarded calls from: * Customer telephone system * Direct central office (Business or Centrex lines) * 800 Service lines Vendor Response Requirement Confirm support for each forwarding requirement.

Cisco Response: Cisco Unity complies. 7.2.2. Disconnect Detection The system should detect that a caller has hung up and immediately disconnect and restore the line to service. Vendor Response Requirement Confirm support for this operation.

Cisco Response: Cisco Unity complies. 7.2.3. Station Dialing



In addition to the menu/route, callers may access an individual station either through the input of the extension number or the input of the called party's last name. A total of 2,000 names plus 100 extension numbers will be possible. Vendor Response Requirement Confirm support for this operation.

Cisco Response: Cisco Unity complies. 7.2.4 Answer Announcement Individual, personalized announcements of 15-30 seconds for each mailbox user will be possible. A user's dictated answer message will only occupy the number of seconds dictated, with the remainder to be pooled so as to be available to: 1) all other mailbox owners; and, 2) for message taking. A system announcement of up to 30 seconds will be possible and also will be available in the event of switching system failure. It will be possible for the mailbox owner to input separate greetings for calls received internally or externally on the system. It will be possible for several individuals to share the same mailbox extension number. A caller reaching such a mailbox will be able to select between individual mailboxes. Vendor Response Requirement Confirm support for these operations.

Cisco Response: Cisco Unity complies. 7.2.5 DTMF Signaling The system will be capable of receiving and generating standard DTMF tone signaling. Vendor Response Requirement Confirm support for this feature.

Cisco Response: Cisco Unity complies. 7.2.6 Greeting Voice mail calls will be answered on the first ring and be time- and date-stamped. Vendor Response Requirement Confirm support for this feature.

Cisco Response: Cisco Unity complies.



7.2.7 Escape A caller reaching the voice mail system will have the ability to re-route to an extension by dialing up to five digits or the operator by dialing "0" before or after leaving a message. It will not be possible for a caller reconnected to the telephone system to be connected to the public network. Vendor Response Requirement Confirm support for this feature.

Cisco Response: Cisco Unity complies. 7.2.8 Trunk Access It will be impossible for a caller passing through the attendant to reach an outside line. Vendor Response Requirement Confirm support for this feature.

Cisco Response: Cisco Unity complies. 7.2.9 Distribution Lists The system will contain a minimum of 80 distribution lists of at least 25 names each plus "all broadcast." Vendor Response Requirement Confirm support for this feature.

Cisco Response: Cisco Unity complies. 7.2.10 Message Forwarding Messages may be forwarded to single or multiple destinations with or without introductory comments. Vendor Response Requirement Confirm support for this feature.

Cisco Response: Cisco Unity complies. 7.2.11 Audit Trail It will be possible for a user to designate a necessary written record of message destination, input time and receipt. This audit trail will be printed on the administrative console together with daily reports. Vendor Response Requirement Confirm support for this feature.

Cisco Response: Subscriber Message Activity Report can be run to accommodate an audit trail, this report can be in a .CSV or HTML file format



to be printed at a later time. The fields available in the report are shown below:

7.2.12 Message Indication The receipt of a message in a mailbox will cause a message-waiting lamp or "stutter" dial tone upon lifting of the station handset to indicate a message-waiting condition. Vendor Response Requirement Confirm support for this feature.

Cisco Response: Cisco Unity complies. 7.2.13 Identification Code Users accessing the system will input a discrete six-digit identification code which will be positively validated prior to access to their mailbox. Identification codes may be changed by mailbox owner.



Vendor Response Requirement Confirm support for this feature.

Cisco Response: Cisco Unity complies. 7.2.14 Message Recovery The mailbox owner accessing the mailbox will be automatically told how many new messages have been received since last access and how many saved messages exist. Upon accessing the messages, the subscriber will have the choice of deleting, skipping or saving a message. Saved messages may only be deleted by the subscriber or by the system administrator. Vendor Response Requirement Confirm support for this feature.

Cisco Response: Cisco Unity complies. 7.2.15 Message Reply A mailbox owner may respond to a message input by another system mailbox owner by simply depressing a single key. Vendor Response Requirement Confirm support for this feature.

Cisco Response: Cisco Unity complies. 7.2.16 Message Review It will be possible for a user to review and edit either an announcement or input a message. Vendor Response Requirement Confirm support for this feature.

Cisco Response: Cisco Unity complies. 7.2.17 User Controls A user accessing their mailbox will be capable of the following control functions: 1. Playback messages 2. Skip to next message 3. Cancel review 4. Replay last message 5. Replay faster or slower 6. Pause 7. Append information 8. Forward message (to mailbox or list) 9. Create new answer announcement



10. Increase play-back volume Vendor Response Requirement Confirm support for this feature. Indicate if any function is not supported.

Cisco Response: Cisco Unity complies. 7.2.18 System Management Console The system will be equipped with a CRT and printer to provide system management functions. The administrative programs and traffic information secured will be possible during system operation. Traffic reports will be available on customer demand or automatically on a pre-programmed basis in quarter, half or one hour time frames or daily and weekly. At a minimum, they will indicate the following: 1. Storage space used for announcements or information mailboxes. 2. Storage space used for messages. 3. Maximum storage space used during the interval. Vendor Response Requirement Confirm support for this feature. Indicate if any requirement is not supported.

Cisco Response: Cisco Unity complies. 7.2.19 Traffic Reports Traffic reports will be available on customer demand or automatically on a pre-programmed basis in quarter, half or one hour time frames or daily and weekly. At a minimum, they will indicate the following: 1. Storage space used for announcements 2. Total calls answered 3. Total calls routed to station 4. Total calls routed to default 5. Total calls abandoned 6. CCS use and call count by input Vendor Response Requirement Confirm support for this feature. Indicate if any requirement is not supported

Cisco Response: Cisco Unity has several reports that can be run on customer demand, scheduled reports are not available. The following reports combined will address the requirements listed above, these reports include Subscriber Message Activity Report, Call Handler Traffic Report, Port Usage Report all combined cover the request above.

7.2.20 System Changeability It will be possible for the system administrator to add and/or delete



mailboxes, change general recordings and perform other administrative duties while the system is in operation. Vendor Response Requirement Confirm support for this feature

Cisco Response: Cisco Unity complies. 7.3.0 Networking VoiceCon plans on networking it new HQ messaging system to other VoiceCon locations equipped with messaging systems. 7.3.1 AMIS The proposed messaging system should support AMIS networking standards. Vendor Response Requirement Confirm support for these features

Cisco Response: Cisco Unity supports AMIS-A networking. 7.3.2 Digital IP Networking The proposed messaging system should support VPIM networking standards. Vendor Response Requirement Briefly describe digital networking capabilities of your proposed messaging system solution. Indicate if VPIM is supported.

Cisco Response: Cisco Unity digital networking allows messaging among multiple Unity servers that access the same subscriber directory. It also provides the means to transfer calls from the automated attendant or directory assistance to subscribers who are not associated with the local server. VPIM support provides digital interoperability with traditional voice mail systems, providing advanced interoperability features, faster message delivery, and lower cost with more efficient message transmission when compared to AMIS message interoperability.

7.4 Integrated Messaging Application Vendor Response Requirement



Briefly describe how the proposed voice messaging system is to be integrated with VoiceCon’s text messaging system, based on a MS Exchange server, to provide unified messaging system functionality. Station users must be able to view and access all messages (voice, text, fax) from their PC display monitor. Email text messages must be accessable from a telephone using text-to-speech conversion.

Cisco Response: Cisco Unity integrates seamlessly with your Microsoft Outlook E-Mail client to make handling all your messages—E-Mail, voice, and fax—easy and convenient, whether you are in the office or on the road. An intuitively designed interface makes it easy to access E-Mail, voice, and fax messages from your desktop PC. Icons provide simple visual descriptions of each message type and because every message is delivered to one inbox, you can see the number, type, and status of all your communications at a single glance. You may also reply to, forward, and save your messages—regardless of media type—in public or personal Microsoft Exchange or Microsoft Outlook folders with just a click of the mouse.

With Cisco Unity's text-to-speech capability, you get information about all your messages—and even hear the text portion of E-Mail messages—over the telephone. You can then respond with a voice message and, depending on the capabilities of your fax server, print E-Mail, attachments, and incoming faxes on a nearby fax machine. The picture below illustrates the user interface for Unified Messaging.



8.0.0 Contact Center (Informational, only) VoiceCon has future plans to install and operate a mid-size contact center solution across its three HQ locations. The contact center would integrate incoming voice, email, and web contacts from customers, and also support outgoing voice calls to potential customers. It is anticipated that the contact center will require 50 multifunction agents positions, and 5 supervisor positions. The contact center features and functions are NOT to be included in the configuration or pricing proposal. 8.1.0 Incoming Voice Call Center The voice contact center solution should support call prompting, detailed call screening, and intelligent call routing capabilities. Agent groups should be both fixed and virtual based on skill profiles of the agents. Client/server CTI applications must be supported at all agent desks. Agent group assignments must be able to be distributed across the three HQ locations. The system should be designed to minimize agent requirements and call waiting times. Realtime supervisor reports and detailed historical reporting is required. Vendor Response Requirement Briefly describe you’re the system architecture of your incoming voice call center solution to satisfy VoiceCon’s basic requirements (see below). Include specific information about the system design architecture of your solution (hardware and software requirements), and specific capacity parameters for agents, supervisors, groups, announcements, queue slots, trunks and trunk groups, et al.

Cisco Response: Read and Understood. Cisco offers two contact center product offerings: 1) Cisco Unified Contact Center Express Edition and 2) Cisco Unified Contact Center Enterprise Edition. The highlights of these two editions are outlined briefly below:

Cisco Unified Contact Center Express Edition provides an integrated ACD, IVR, and CTI solution for up to 300 agents and 300 IVR ports. Unified Contact Center Express Edition offers three licensing options -- Standard, Enhanced, and Premium. Key features for Unified Contact Center Express Edition include:

• Contact-center-in-a-box: Fully integrated ACD, IVR, and CTI • Multi-site, virtual contact center • CRM customer profile routing and prioritized queuing • Agent skills and competency based routing • TOD, DOW, ANI, or DNIS routing • Expected delay and position in queue announcements • Queuing treatment based upon CRM customer profile • Voice mail capture and routing with original screen pop • Scheduled callbacks with original screen pop • Automatic callback of abandoned calls with original screen pop • Web callback with screen pop



• Robust Cisco Agent Desktop (CAD) that provides hot desking, agent state control, call control, call recording, queue statistics, agent statistics, text chat with other agents and supervisor, wrap-up state, workflow buttons, and CRM application screen pop

• Robust Cisco Supervisor Desktop (CSD) that provides supervisors the ability to view/change agent states, text chat with other agents and supervisors, silent monitor, barge-in, intercept, record, and record playback.

• ODBC Compliant CRM database integration support for IVR self service application

• Integrated VoiceXML browser to rendering VoiceXML applications.

• Integration with ASR and TTS services via MRCP Cradle-to-grave Contact Call Detail Records (CCDRs)

• Real-time and Historical Reporting • Support for internationalization; supported localizations include

American and UK English, French and Canadian French, German, Spanish, Italian, Japanese, Chinese and others

• Softphone support option • Browser based administration • High Availability option • Integration with traditional ACD and IVR systems via the Cisco

ICM that provides load balancing, cross-system call data passing, and consolidated multi-system reporting

The average list price cost per Unified Contact Center Express agent is between US$700 and US$1500 in typical configurations. Cisco Unified Contact Center Enterprise Edition also provides an integrated ACD, IVR, and CTI solution. Unified Contact Center Enterprise can support over 10,000 IVR ports and agents. In addition, Unified Contact Center Enterprise offers a robust set of multi-channel features for handling web callback requests, E-Mails, text chat requests. Web collaboration can also be added to any voice or text chat contact to improve communication effectiveness, agent productivity, and customer satisfaction. Unified Contact Center Enterprise also offers a comprehensive outbound option that provides predictive, progressive, and preview dialing modes. Cisco Unified Contact Center Enterprise Edition segments customers, monitors resource availability, and delivers each contact to the most appropriate resource anywhere in the enterprise. To complete this transaction, the software profiles each customer using contact-related data such as dialed number and calling line ID, caller-entered digits, data submitted on a Web form, and information obtained from a customer profile database lookup. Simultaneously, the system monitors the resources available in the contact center to meet customer needs, including agent skills and availability, interactive-voice-response (IVR) status, queue lengths, etc.



This combination of customer and contact center data is processed through user-defined routing scripts that graphically reflect a company’s business rules—enabling Cisco Unified Contact Center Enterprise to route each contact to the optimum resource anywhere in the enterprise. Wherever an agent is based, the system delivers a uniquely rich set of call event and customer profile data to the targeted desktop as a contact arrives, personalizing service and maximizing efficiency. Throughout the process, carrier-class, distributed fault tolerance from the network to the desktop ensures uninterrupted operation. Cisco Unified Contact Center Enterprise Edition is priced as a mid-market to high-end solution. The average list price per agent ranges from US$900 to US$2000.

Proposed Solution Based on VoiceCon’s multi-channel requirement, outbound dialing requirement, and the requirement to support a call center spanning three headquarters locations, Cisco would propose the IP Contact Center (IPCC) Enterprise Edition. All of the components needed to meet the RFP requirements would be provided by Cisco. Unified Contact Center Enterprise runs on separate servers that communicate with Cisco Unified CallManager server via JTAPI. Cisco Unified Contact Center Enterprise Edition delivers intelligent call routing, network-to-desktop CTI, and multimedia contact management to contact center agents over an IP network. By combining software ACD functionality with IP telephony in a unified solution, Unified Contact Center Enterprise enables companies to rapidly deploy a distributed customer contact infrastructure to support their global e-sales and e-service initiatives—an infrastructure that we call a customer interaction network. Cisco Unified Contact Center Enterprise delivers an integrated suite of proven solutions that combine Cisco IP telephony and contact center solutions. It utilizes a company’s existing IP network, thus optimizing investments in wide area network (WAN) infrastructure and lowering administrative expenses. Moreover, this IP-centric architecture allows your business to easily extend the boundaries of the contact center enterprise to include branch offices and knowledge workers. Cisco Unified Contact Center Enterprise is designed for implementation in both single-site and multi-site contact centers as well as service provider hosting environments. Specific capabilities include:

• Intelligent multi-channel contact routing • Network-to-desktop computer telephony integration (CTI) • Interactive voice response (IVR) integration • Comprehensive outbound dialing, and • Real-time and historical reporting.

Cisco Unified Contact Center Enterprise is a a scalable, distributed computing solution comprised of the following components:



Cisco VoIP Gateways

The Cisco VoIP gateway provides a connection path between the PSTN and the Cisco IP communications network that converts analog and digital voice into IP packets.

Cisco Unified IP Phones or IP Communicator (Softphone)

IPCC agents utilize Cisco Unified IP Phones or IP Communicator. Cisco Unified IP Phones are full-featured, second-generation voice instruments that use IP transport technology to consolidate data and voice into a single network infrastructure—including a single cable plant, a single switched Ethernet fabric for campus or branch offices, and unified systems for operations, administration, and management. Cisco IP Communicator provides a softphone option for mobile agents

Cisco Unified CallManager

Cisco Unified CallManager combined with Cisco VoIP gateways, Cisco Unified IP Phones, and a Cisco data network provides an enterprise class, feature rich, IP PBX. Cisco Unified CallManager software runs on Cisco MCS servers or approved servers from HP or IBM. Cisco Unified CallManager provides call setup, teardown, and call control (i.e., call processing).

Cisco Intelligent Contact Management (ICM)

Cisco ICM is the central brain for an Unified Contact Center Enterprise deployment and runs on Cisco MCS servers or approved servers from HP or IBM. The ICM provides the multi-channel ACD, CTI, and outbound dialing functions, but it also controls the overall customer interaction including usage of IVR applications for self service and queuing treatment. The ICM software monitors agent states, makes agent selections, controls call queuing, and provides browser based consolidated reporting for all customer interactions. The ICM software is also responsible for managing the CTI interface to an agent desktop application for screen pop and 3rd party call control and monitoring. The ICM is also responsible for managing outbound dialing campaigns and blending outbound calling with inbound calling. The ICM also provides integration to traditional ACD and IVR systems to allow intelligent load balancing, to allow call data to be passed between systems, and to allow multi-system consolidated reporting. The ICM software also provides an option to integrate with carrier networks call routing intelligence to provide PSTN pre-routing.

Cisco Unified IP IVR

The Cisco Unified IP IVR provides self service applications and also functions as a queuing point to allow intelligent call treatment while callers are in queue. Cisco Unified IP IVR software runs on Cisco MCS servers or approved servers from HP and IBM. These call treatment messages can include predefined greetings as well as dynamically generated content to convey queue status.



Cisco E-Mail Manager (CeM)

The Cisco E-Mail Manager is an optional Unified Contact Center Enterprise component that functions as a POP mail server and as a contact routing client to the ICM. CeM software runs on Cisco MCS servers or approved servers from HP and IBM. When the CeM receives a new E-Mail, it parses the E-Mail to determine the intent and then passes contact details to the ICM so that the ICM can make a routing decision for that contact. CeM has the ability provide an automated reply to customers and also has the ability to deliver a pre-defined template to the agent desktop in an effort to reduce agent handle time. CeM also provides complete archiving of the customer E-Mail interactions.

Cisco Web Collaboration Option

The Cisco Web Collaboration Option is an optional Unified Contact Center Enterprise component that provides web callback, text chat, and web collaboration functions. This software runs on Cisco MCS servers or approved HP or IBM servers. It functions as a web server and provides options to web customers for web callback or text chat interactions. Cisco Web Collaboration Option passes contact details to the ICM so that the ICM can make a routing decision for that contact. Any customer data collected can be used by the ICM routing script and sent to the agent desktop for screen pop. After either a web callback or text chat session has been established, the agent and the customer can add a web collaboration session to add a visual component to the customer interaction.

Cisco Outbound Option

The Cisco Outbound Option is an optional Unified Contact Center Enterprise component that provides preview, progressive, and predictive dialing modes. The dialer software runs on Cisco MCS servers or approved servers from HP or IBM. This software based dialer has the ability to blend outbound calling with inbound calling, manage multi-site campaigns, and perform call progress detection (for answering machines). Via the ICM browser based reporting, consolidated inbound and outbound reporting is offered.



IPCC Capabilities and Benefits: Routing Capabilities and Benefits

Cisco Pre-Routing and Post-Routing Functionality The Cisco Pre-Routing® function makes routing decisions for each call while it is still in the carrier's network, enabling the Cisco platform to effectively segment customers, balance calls across the enterprise, and deliver each contact to the best enterprise resource the first time.

The Cisco Post-Routing® function provides the intelligent distribution of contacts already connected to a peripheral (ACD, private branch exchange [PBX], IVR, or Web or E-Mail server) in a company's private network. When a contact requires redirection, Cisco Unified Contact Center Enterprise applies business logic instructing the peripheral to send the contact to the best available enterprise resource. For contacts flowing between sites, among agents, skill groups, or IVRs, Post-Routing optimizes each customer's interaction by retaining data collected during the Pre-Routing function and applying it to the Post-Routing function—eliminating the need for the end customer to restate any previously collected information.

Customer Profile Routing Cisco Unified Contact Center Enterprise Edition extends the sources of data available for making contact routing decisions and for populating agent desktop applications. For instance, Cisco Unified Contact Center Enterprise is able to perform a customer profile database lookup during routing to more effectively segment customers and determine the optimal destination for each contact. Moreover, information obtained from customer relationship management (CRM) packages can be used to even more precisely match customers with agents and expand the data available to screen pop applications.

Cisco Web Collaboration Option The Cisco Web Collaboration Option enables your contact center agents to respond immediately to questions from customers using your Web site backed by Web pages and other Web-hosted content. Agents may also use Web collaboration to help a customer via simultaneous voice-and-visual interaction. Web collaboration allows contact center agents and customers to share Web pages, collaboratively complete online forms, and share any Windows desktop application using nothing more than a Web browser. Cisco Unified Contact Center Enterprise will route customer requests from your Web site utilizing the same call routing logic used for voice calls. By facilitating effective, personalized assistance designed to greatly enhance the customer experience, Cisco Web Collaboration Option is an ideal solution for both sales- and service-oriented contact centers.

Cisco E-Mail Manager Option

The Cisco E-Mail Manager Option is a comprehensive, enterprise-class solution for managing high volumes of customer E-Mail inquiries submitted to your company mailboxes or Web site. Based on customizable business rules, the Cisco E-Mail Manager Option accelerates the response



process by automatically directing messages to the right agent or support team, categorizing and prioritizing messages, suggesting relevant response templates, and, if desired, sending automated replies. Quick, accurate turnaround of inquiries translates to strengthened customer relationships, and added value and efficiency in the contact center.

Cisco Outbound Option

The Cisco Outbound Option and its combination of outbound dialing modes complements the powerful inbound call handling capability of the Cisco Unified Contact Center Enterprise platform with a robust outbound call management solution. Campaigns can be built to utilize predictive, progressive, or preview dialing, which is integrated with inbound calls and compliant with contact center service levels to offer a powerful blended solution. Agents can be allocated to handle pure inbound, pure outbound, or both inbound and outbound contacts, offering an effective way to increase resource utilization in a contact center.

Agent Capabilities and Benefits

Cisco Computer Telephony Integration Option

Cisco Unified Contact Center Enterprise software enables companies to deploy a complete network-to-desktop CTI strategy, including comprehensive functionality at the agent's workstation. Cisco Unified Contact Center Enterprise solutions deliver a uniquely rich set of data to business applications, providing enterprise-wide call-event and customer-profile information to a targeted agent's desktop. Cisco Unified Contact Center Enterprise sets a new standard for true enterprise-wide, network-to-desktop CTI with minimal custom development or systems integration, enabling an organization to implement CTI quickly and cost-effectively.

Universal Queue Cisco Unified Contact Center Enterprise coordinates an agent's ability to work on multiple tasks from various channels while allowing your agents to be interrupted with high priority tasks as required. For instance, if an agent is assisting a customer using text chat, another text chat request could be delivered to that agent, increasing the agent's productivity. Agents also may be delivered a task of a different channel type than their active task. For example, an agent may be responding to a customer's E-Mail when a voice call arrives. Cisco Unified Contact Center Enterprise can route the voice call to the agent, interrupt the agent's work on the E-Mail, allowing the agent to handle the real-time voice call, and then return to the E-Mail when the voice call is complete. In this way, Cisco Unified Contact Center Enterprise can optimize your agent's activities, ensuring the highest level of customer service with the resources available.

Unique to universal queue is the ability to accurately report on an agent's tasks and activities. Cisco Unified Contact Center Enterprise provides real-time status and historical reporting of universal queue and task interruptability. Real-time displays accurately depict the agent's current task and time associated. Historical reports track the cumulative time associated with these tasks, omitting the time the agent's focus was diverted to another routed task.



Remote Agent Support

Remote agent support extends the Cisco Unified Contact Center Enterprise environment by providing CTI, contact distribution, and reporting capabilities to remote agents. In addition to skills-based routing, Cisco Unified Contact Center Enterprise provides softphone, screen pop, and third-party call control functionality for remote agents—delivering a single, network-to-desktop CTI solution for the enterprise. By incorporating agents outside of the physical location of a contact center, the Cisco Unified Contact Center Enterprise solution enables companies to better utilize existing and on-demand resources and fully extend CTI functionality across an entire enterprise, regardless of agent location.

Call Treatment Capabilities and Benefits

IVR Integration Cisco Unified Contact Center Enterprise supports call queuing and treatment on a premises- or network-based IVR. This intelligent call treatment device allows your contact center to provide dynamic information to your callers while they are queued for an agent. This dynamic information could consist of the number of calls in queue and expected wait time or product or service messages tailored to your callers' needs. For instance, callers might be prompted for their account code and offered options to be routed to the service desk. The Cisco Unified Contact Center Enterprise accesses the customers' data files and, while waiting for an agent, callers hear service information related to their product, options, and service history. The result is better-informed callers, reduced agent talk time, and increased up-sell and cross-sell opportunities.

Management Capabilities and Benefits

Supervisory Features

Cisco Unified Contact Center Enterprise Edition allows supervisors to view agent states and call information, send text chat messages to agents, barge-in and intercept calls and, depending on the configuration of the desktop, record conversations and silently monitor agent calls. These features add value to the supervisors' role in the contact center and help them effectively manage their team.

With supervisor and agent chat capabilities, supervisors can send text messages to agents participating in a call. This allows supervisors to coach agents on cross-sell and up-sell opportunities and enables agents to resolve customer issues without putting the customer on hold to consult with a supervisor or another agent. Barge-in allows supervisors to interrupt an agent's call—creating a three-way conference. The supervisor can then interact with both the caller and the agent to help resolve a caller issue. A supervisor can remove the agent from a call using the intercept feature, allowing the supervisor and caller to complete the call on their own while freeing the agent to handle another customer request.

Supervisors also can change an agent's state from their desktop. New agents commonly have difficulty remembering to make themselves available to take calls after a break, or to log out when away from their workstation for an extended period. A Cisco Unified Contact Center



Enterprise supervisor can easily remedy these situations with a simple click of the mouse—logging out missing agents or making unintentionally idle agents ready to take calls.

Finally, advanced features such as call recording and silent monitor provide supervisors with the tools they need to effectively manage their team. Call recording allows supervisors to initiate recording of an agent conversation and play it back at a later time. These recordings can then be used in quality assurance or training sessions, providing audible proof that a high standard is consistently being met. Equally important is the silent monitor feature, which provides supervisors the ability to select an agent and monitor his or her active calls in real time. This function improves the supervisors' knowledge of contact center activities, allowing them to gauge quality levels and pinpoint potential training needs.

Administration Streamlined multi-channel administration allows managers to set up a Cisco Unified Contact Center Enterprise agent once through a single user interface. The intuitive, user-friendly interface allows agents to be set up to handle voice, Web, chat, and E-Mail contacts, depending on their assigned skill sets. The administrative workstation is the user interface into the Cisco Unified Contact Center Enterprise environment, enabling system managers, administrators, and supervisors to define, modify, or view routing scripts; manage the system configuration; monitor contact center performance; define and request reports; and ensure system security. This one user interface provides enterprise-wide control across the single- or multi-site contact center.

Reporting The Cisco Unified Contact Center Enterprise solution provides a collection of real-time and historical data necessary for mission-critical contact center reporting. Cisco Unified Contact Center Enterprise reporting provides accurate and timely browser based reports on contact center activity, enabling managers to make informed decisions regarding staffing levels, contact handling procedures, and technology investments. Standard reporting templates provide "out-of-the-box" functionality for common reporting needs. Custom reports extend the standard reporting package to meet your specific reporting needs. Furthermore, the open software architecture of Cisco Unified Contact Center Enterprise allows for the consolidation of timely and accurate information from the Internet, carrier networks, ACDs, IVRs, Web servers, databases, business applications, and other resources, creating a complete view of the contact center enterprise.

System Capabilities and Benefits

Open Systems

Cisco Unified Contact Center Enterprise software takes full advantage of industry-standard software enabling customers to reap the benefits of world-class software functionality at a modest hardware cost. Moreover, the open architecture of the system, which includes an Open Database Connectivity (ODBC)-compliant database, an open Computer Supported



Telecommunications Applications (CSTA) switch interface, and Java and ActiveX interfaces for CTI applications, integrates existing contact center solutions, preserves investments in traditional systems, and provides a platform for future applications.

Scalability Cisco Unified Contact Center Enterprise software supports both co-resident and nodal architectures and can scale to accommodate thousands of agents at multiple sites depending on the contact center requirements. If your company's use of the Cisco Unified Contact Center Enterprise platform grows to span many locations, the system is fundamentally architected to accommodate your changing environment while protecting your initial investment.

Distributed Fault Tolerance Cisco knows that the handling of customer contacts is a mission-critical business function. From the network to the desktop, all Cisco Unified Contact Center Enterprise software components and external application links provide carrier-class, distributed fault tolerance at both the hardware and software levels, with real-time application failover capabilities. Self-diagnostics and self-healing features allow the system to automatically take advantage of redundant components when required; the system is resilient to hardware component failures, communications network failures, and asynchronous software errors. Cisco Unified Contact Center Enterprise software also includes a Simple Network Management Protocol (SNMP) feed for integration into a corporate-wide fault-management system.

Multi-carrier, Multi-vendor Capabilities The Cisco solution helps customers meet business objectives without the limitations of proprietary or custom solutions. The open architecture of the Cisco Unified Contact Center Enterprise software supports a heterogeneous environment of carrier networks; ACD, private branch exchange (PBX), IVR, Web, and E-Mail platforms; and complementary software applications—enhancing the value of existing investments while supplying advanced functionality that traditional offerings do not provide (ask your representative for a complete list of current network and platform support).

Cisco Unified Contact Center Enterprise support for legacy ACD and IVR platforms provides a unique migration path to a full Cisco IP Contact Center. By integrating to your existing ACD and IVR platforms, Cisco Unified Contact Center Enterprise allows users to fully realize the investments in these products while beginning the migration to IP. By combining previously heterogeneous applications, the Cisco Unified Contact Center Enterprise solution also permits normalized, consolidated reporting and enables a company to apply consistent performance standards across the enterprise. As these devices are fully depreciated, their functionality is replaced by Cisco Unified Contact Center Enterprise to complete the migration.

Finally, integration with best-of-class applications for CRM, agent scheduling, workflow management, voice recording, and other activities allows a company



to address unique business requirements while maintaining an enterprise view of contact center performance.

8.1.1 Basic Call Control Capabilities At a minimum the proposed solution must be able to provide call control based on: • ANI/DNIS • call volumes • performance criteria • priority queuing Vendor Response Requirement Briefly describe the call control methodology used by your system that analyzes, routes, and queues calls based on each of the criteria.

Cisco Response: Cisco Complies. DNIS, CED, ANI, day, date, account number etc. Calls can be routed based on a multitude of dynamic variables or rules that are defined by the network provider or end-user including; dialed number, CLID, location of the caller, caller-entered digits, customer database lookup, agent availability, agent skills, time of day, day of week, scheduled workforce, workload distributions, or the cost of the phone call. ICM software monitors the real-time availability and performance of the agents in a contact center via the ICM system’s Peripheral Gateway (PG), and utilizes that information in controlling the routing of calls in the carrier’s network.

8.1.2 Advanced Call Control Capabilities As an option the proposed solution must be able to provide call control based on: • agent skills • customer preference • inbound and outbound call levels • multi-media Vendor Response Requirement Briefly describe your system’s call control methodology that analyzes, routes, and queues calls based on each of the criteria.

Cisco Response: Cisco Complies Defining Routing Logic Unified Contact Center Enterprise Edition’s Script Editor provides a visual, object-oriented environment for defining call routing logic. In appearance, a script resembles a programming flowchart. The call routing logic is put together by selecting the appropriate call routing objects from a palette of drag-and-drop objects and placing them in a script window. You can then define the relationship between objects with interconnecting lines. The call routing script shown below is an example of a help desk application



where the business goals are to:

• Distribute the call volume in accordance with the staffing at each location. • Ensure customer satisfaction by balancing the load across all the sites in

case any of the locations are unable to meet their service level objectives.

Example Call Routing Script The graphical nature of the Script Editor helps you to understand the call flow for a given script based on visual inspection. A script usually has several branches that can be followed depending on current conditions at the contact centers. In the script, a call might be routed to a particular skill group or service within one of the target sets.

Verifying Routing Logic Some call vector and routing tree programs make it difficult to determine whether the routing scripts are truly sending calls to the appropriate destinations on a call-by-call basis. The ICM software Script Editor overcomes these limitations by displaying call volumes and percentages of calls in the context of the script that is active.



Routing Script in Monitor Mode The script monitoring capability allows the determination of precisely how many calls took a given path in the ICM software script. You can then compare that data to the conditions that existed in the network during the same interval and expose anomalous behavior that exhibits itself only under specific load conditions. With the Script Editor’s advanced monitoring and editing capabilities, you can identify a potential call load problem, isolate it, make a script change, and rapidly introduce the change into the system. Routing scripts let you implement the business goals of your organization by determining the routing decisions ICM software makes under certain circumstances. For each incoming call, ICM software uses the call information provided by the routing client (for example, Dialed Number or Caller Entered Digits). It then applies a series of scripts along with its knowledge of contact center activities until it determines a destination for the call. The call information provided by the network routing client is its call type. The call destination, which is derived through the use of the scripting algorithms, is typically the site most likely to give the caller fast, helpful service. ICM software refers to the destination of a call as the routing target. Segmenting Callers Cisco ICM software provides a mechanism for logically segmenting callers based on the type of service request they are making. For example, a



distinction can be made between new and existing accounts making new product inquiries, or callers from France can be segmented from those calling from Germany. When the ICM system receives a call routing request, it can examine any information available about the call, or caller, and use that information to classify the call as a particular call type.

The elements that can be used to determine a call type include:

• Dialed number (DN). The telephone number that the caller dialed (for example, 800-555-1212).

• Caller-entered digits (CED). Additional information entered, or spoken, by the caller in response to prompts. This can be as simple as “Press 1 for Service, Press 2 for Sales” or it can be a request to collect more caller specific information such as an account number, personal identification number, trouble ticket number, etc.

• Calling line ID (CLID). The caller’s phone number, or billing telephone number, also referred to as Automatic Number Identification (ANI). This can be expressed as a 10-digit phone number, an area code, an area code with local exchange, or a geographical region composed of several area codes.

Any combination of these elements can be used to define a call type, as shown in the sample screen below.

Call Type Directory Entry Any call types that you define can be listed in the Call Type Directory dialog box, shown below.



Call Type Directory Scheduling Routing Logic Cisco ICM software provides flexibility in scheduling when a given call routing script is used to route calls of a certain type. Some scripts might be active at all times; others might be active only during certain hours of the day or certain days of the week, month or year. You may even define a script to run only on a specific date during a specific period of time. Standard Routing Targets After determining the call type, ICM software runs one or more scripts to select a routing target for the call. The ICM system can route a call to any of the following target objects:

• An agent. An individual at a specific contact center that is able to handle this call.

• A skill group. A group of individuals at a specific contact center that are able to handle this call. The local switch at the contact center is responsible for eventually picking an agent.

• A service. The type of work or information that this caller needs and a specific contact center that can provide it. The local switch at the contact center is responsible for eventually picking a specific agent to handle the call.

• An enterprise skill group. A collection of individual skill groups from several contact centers. ICM software then further qualifies the target to determine which site has an agent that can best provide these skills.

• An enterprise service. A collection of individual services from several contact centers. ICM software then further qualifies the target to determine which site can best provide this service.

• An announcement. A verbal message to the caller. After the caller hears the message, the call may be terminated.

• Busy. Play a busy signal for the caller. This effectively terminates the call. • Ring. Play an unanswered ring for the caller. This effectively terminates

the call.

Queuing Cisco ICM software’s IVR integration supports call queuing at a network- or premises-based IVR. Using the ICM system’s Service Control Interface, a call can be directed to an IVR queue when no appropriately skilled resource is available. Standard call treatments such as announcements, Play Music or Collect Digits can be applied to the call as it is held in queue. When an agent in a targeted skill group becomes available anywhere within the enterprise, ICM software directs the IVR to release the call. Queuing of calls is handled by the ICM with the help of a queue point. The ICM logically queues the call – it associates the call with one or more skill groups and monitors their availability. Physically the call is sent to a queue point, in most cases an IVR for call treatment messages. These messages may be:



• Static Announcements: “All of our agents are busy . . . ” • Dynamic Announcements: “There are 12 calls in queue. Your expected

wait time is 15 minutes” • Prompts for caller data: “Please enter your 6 digit account code.” • Prompts for route options: “To leave a message press 1, to select form a

list of frequently asked questions . . .”

The advantage of using an IVR for call treatment over a voice announcer which most PBXs use, is that it is an intelligent device which can provide dynamic content to queued callers. Through an interchange of data between the ICM and IVR (caller entered digits, ANI, DNIS, etc.) the caller is recognized and call treatment messages that are pertinent to that caller are played. For example, we know who the caller is based on the account code they entered, we know they are calling the service line for product support, therefore, the IVR is commanded to play a script that provides the caller with service options mixed with music. It’s through the ICM’s Open IVR Interface SCI (Service Control Interface), that the ICM can tell the IVR which messages to play and remove the call from the IVR when an agent becomes available. An IVR that supports this interface is required to queue calls in the IPCC. Vendors who support this interface are: • InterVoice • Nortel (Periphonics) • Lucent/Avaya (Conversant) • Edify • Cisco Internet Service Node (ISN) and Cisco IP-IVR The Cisco IP-IVR and ISN support the SCI and can be a fully integrated component of the IPCC.

8.1.3 Caller Notification of Wait Time The proposed solution must be able to notify callers of expected wait times and “place” in queue and support information collection (such as an automated attendant feature) using “internal” hardware and software. Vendor Response Requirement Describe how the application calculates wait time and any optional hardware or software required. Include a statement addressing if the announcement of wait time has an impact on a caller’s state in queue?

Cisco Response: Cisco IP–IVR is an option for an Unified Contact Center Enterprise deployment, providing call treatment to calls in queue. Cisco IP-IVR is easy to install, configure, and manage. Cisco IP-IVR allows callers to select routing options, providing easy access to multiple agent skill groups, extensions, or announcements, either before or after routing. Cisco IP-IVR call-treatment messages can be static, prerecorded announcements or dynamic announcements tailored to specific caller interests. Like a Web site that displays content based on a user's previous visits, Cisco IP-IVR can provide dynamic content to queued callers, delivering



unique messages tailored to each caller's needs such as, the route selected, the caller's place in the queue, or other associated values.

8.1.4 Transfer to Voice Messaging Application After a configurable time, the caller should be able to transfer to a voice messaging system to leave a callback message. Vendor Response Requirement If the caller chooses to continue waiting rather than hanging up after leaving a message, describe how the call is placed back in queue.

Cisco Response: Cisco complies. Cisco IP-IVR supports a script node that would enable customers to leave a message, rather than waiting in queue. The IP IVR can then generate a phantom call into the Unified Contact Center Enterprise system. The phantom call will be routed based upon agents skills and customer priority. Upon agent selection, their screen is popped with the original call data and the agent is played the customers voice message. A callback request can be tied to this voicemail message if desired.

8.1.5 GUI Administration Tool Supervisors must be able to reconfigure call control and assignments in real time, change priority of multiple calls simultaneously, view details of orphaned calls and retain customized settings regardless of log-on location. The solution must use a GUI administration tool and provide a graphical editor and what-if modeling as standard. Vendor Response Requirement Describe the system’s GUI administration tools.

Cisco Response: Cisco complies. Cisco Unified Contact Center Enterprise Edition has a full suite of GUI-based administrative tools, for call scripting (call control), real-time reporting and management, and historical reporting. See section 8.1.2 for details on the call control scripting environment, including what-if modeling.

8.1.6 Soft Client A soft client agent telephone and supervisor console will be highly desirable for both premises and off-premises locations. Vendor Response Requirement



Describe the soft clients available for agent and supervisor use. The soft client must provide on-line help, ability to reserve calls or change call priority. For proprietary clients, detail minimum hardware and software requirements.

Cisco Response: The Cisco Agent Desktop for the Cisco IP Contact Center (IPCC) solution is a robust computer telephony integration solution for single-site and multi-site, IP-based contact centers that are easy to deploy, configure, and manage. It provides agents and supervisors with powerful tools to increase productivity, improve customer satisfaction, and reduce costs. Intuitive, GUI-based management decreases IT dependency and supports simplified customization, maintenance, and change management. Key features and benefits include:

• Robust CTI screen pop, soft phone with media termination and agent/supervisor coaching capabilities—improves agent productivity, which translates to faster, more efficient customer service and improved customer satisfaction

• Ease of deployment—installs faster and easier than traditional CTI technologies

• GUI-based management for ease of customization, maintenance and change management—decreases IT dependency

• Supervisory features allow you to view real-time statistics, monitor and coach agents, barge-in, intercept and record active agent calls when necessary

• Allows companies to rapidly deploy CTI at new locations and migrate to IP at their own pace

Supervisor Desktop - The Cisco Agent Desktop includes software that allows the supervisor to view agent states, call information, send text messages to agents, record conversations and provide advanced monitoring functions. This desktop application empowers the supervisor to more effectively manage his or her agent team.

Supervisor Desktop:



Specific capabilities include the following:

Real-time Visual Agent Call Status • Displays agent’s name, extension, agent state, whether an inbound or outbound call, duration of call and if call is being recorded from the supervisor’s desktop • Supervisors can view real time statistics for individual agents, for teams and for queues • Assists in staffing solutions and provides a fast and easy way to monitor contact center’s activities • Supervisor can change an Agent’s ACD state (“not ready” to “ready”; “ready” to “log out”; etc.) Supervisor/Agent Chat—With the Cisco Agent Desktop’s internal text chat capability, supervisors can send text messages to agents participating in a call. This allows supervisors to coach agents on cross-sell and up-sell opportunities and enables agents to resolve customer issues without putting the customer on hold to consult with a supervisor or another agent. Silent Monitor Function—Supervisors have the ability to select an agent and monitor calls in real-time. This function improves the supervisor’s knowledge of contact center activities and enables him or her to gauge quality levels, pinpoint training needs and oversee performance of contact center agents. Barge-in—Supervisors can also barge-in to an agent’s call through real-time monitoring or by responding to a supervisor alert when the agent requires assistance. The supervisor can then interact with both the caller and the agent to help resolve a caller’s issue. Barge-in provides assistance to the agent when escalation to a supervisor is warranted Intercept—A supervisor can remove an agent from a call, allowing the supervisor and caller to complete the call on their own. Coaching – While using the Silent Monitoring feature to listen to an agent’s conversation with a caller, the supervisor may choose to communicate with the agent through the text chat feature. The caller is unaware that this text-based conversation is in progress. Coaching allows the supervisor to provide guidance and support to new agents or those with special needs. Call Recording and Recorded Call Review—Cisco Supervisor Desktop includes a button to initiate recording and also stop recording of an agent conversation. It provides a recorded call reviewer application to select and playback recorded calls. This application is independent of supervisors to allow persons, such as quality assurance or training persons, to review recordings without being logged in as a supervisor. Supervisor Monitor - Real-time monitoring improves the supervisor’s knowledge of contact center activities. It provides a vehicle to gauge quality levels and critique contact center agent performance and allows a supervisor to unobtrusively assist agents with difficult callers. Marquee Message – This feature allows for instant broadcast messages to all agents or groups. It allows for up-to-the-minute information with instantaneous message changes. Conference Information - Provides information regarding agents and customers involved in conference calls.



Supervisor Hot Desking – The Supervisor profile is held in an LDAP directory on a server allowing supervisors to move from one PC to another.

8.1.7 ACD Voice Terminal IP desktop voice terminal instruments will be required for agent positions. Vendor Response Requirement Briefly describe any telephone instruments designed specifically for ACD agents. Include any and all feature/function attributes unique to ACD operations. Provide a photograph of the instrument, if available.

Cisco Unified IP Phone 7961G Cisco Response: Unified IP Phone Agent - The Unified IP Phone Agent allows agents to use the Cisco Unified IP Phone 7961G telephone as an ACD Phone, in place of a CTI Agent Desktop application. Agents can log in and out of the ACD, view ACD state, change states, enter reason codes and wrap data. Supervisors can view agent states, change agent states and monitor agents using Unified IP Phone Agent. For agents using CTI Desktop application, they may use the Unified IP Phone Agent as a back-up agent phone in case of a PC failure. Cisco Unified IP Phone 7961G's have a large graphical display which is capable of providing extended services to the phone. If ANI information is available, it is shown on the phone. If ANI is not available, the display is typically configured to show the trunk number or group. The IPCC Agent Desktop interface will display ANI, DNIS Caller Entered Digits and other selectable data fields. This interface also provides full Screen Pop capabilities to present the agent with a complete view of the caller at the call’s arrival.

8.1.8 Supervisor Real-time Call Handling and Performance Status Supervisor terminals must show, in real time, all logged-on



agents, the status of each agent, caller queue information and thresholds and alarms. Users must be able to customize displays. Vendor Response Requirement Describe the proposed solution's real time supervisor console display capabilities for assisting supervisors with managing the customer interaction center. Include a diagram illustrating two or three screen displays available to the supervisor.

Cisco Response: Cisco complies. Cisco Supervisor Desktop provides a variety of features for supervisors to utilize for agent supervision. These include:

• Real-time statistics • Text chat • Marquee messaging • Make agent ready • Log out agent • Silent monitor • Barge in • Intercept



8.1.9 Agent Display Information Vendor Response Requirement Describe real-time display information provided to agents at their desktop via their hard telephone instrument and the softclient solution.

Cisco Response: Cisco complies. Cisco Agent Desktop provides an intuitive interface for agents, either via the Unified IP Phone (hard phone) or soft client. Key values of the softphone include:

• Minimal Screen Space • Call Control Features - Answer, Release, Dial, Transfer,

Conference • ACD State Control - Log In, Log Out, Ready, Not Ready • Chat with other agents or supervisor • View current agent and skill group statistics • Quick Installation



Key values for the Unified IP Phone-based agent interface include:

Cisco 7960/7940 Unified IP Phones • Log in/out • Ready/not ready



• Supervisor desktop shows agent phone state • Supervisor silent monitoring • Enterprise data display (data pop)

8.2.0 Reporting VoiceCon requires call center system operation reports in various formats. 8.2.1 Statistical and Configuration Reporting VoiceCon requires sophisticated reporting to track and further enhance its CIC operations. Reports must be available on terminal display and paper printout and be able to be downloaded to a PC. The proposed solution must provide open storage capability. Vendor Response Requirement Describe the number of and type of information standard statistical, configuration and audit reports provided.

Cisco Response: IPCC has a pre-configured set of reports, which provide agent and system information (trunk activity must be viewed separately in Unified CallManager - refer to section 3.0 for details). IPCC also supports customized reporting. IPCC has a report job scheduling capability that allows the administrator to designate when specific reports will be generated. Reports can be accessed and printed at any time without affecting the integrity of subsequent data. Real-time reports provide the following essential information, both system-wide and by agent:

• Calls In Progress (talking to agents) • Calls Handled • Average Queue Time • Current Agent State • Time of last Agent State Change • Calls in Queue by Call Type



• Calls in Queue by skill group • Longest Call in Queue by Call Type • Service Level per Call Type • Abandoned Calls by Call Type • Threshold Alarms

Historical reports are also available. By category, these include:

• Agent Activity • Agent Call/Contact Detail • Agents by Name • Skill Group Reporting • Abandoned Calls by Call Type • Abandoned Calls by skill group • Average Speed to Answer by Call Type • Average Handle Time per Agent • Exception Reporting (RNA, Time outs) • Call Type (Pilot Number) Activity • Daily • Weekly • Monthly • Selectable Interval • Service Levels by Call Type • Longest Call in Queue by Call Type

8.2.1 Graphical Reporting The proposed solution must provide graphical reports as a standard feature. Vendor Response Requirement Describe the available graphical reports with your system.

Cisco Response: Cisco complies. All reports available with the IPCC system are available in tabular or graphical format. Sample reports include:





8.2.2 Call-by-Call Reporting The proposed solution must provide call-by-call reporting as an optional feature. Vendor Response Requirement Describe your system’s call-by-call reporting capabilities, if available.

Cisco Response: Cisco complies. Cisco Unified Contact Center Enterprise Edition’s flexible reporting capabilities include the ability to deliver call-by-call reporting. This is a standard report.

8.3.0 Self Service The proposed solution must support self service (e.g., IVR) integration as an option. Callers must be able to retain their place in queue while using IVR features Vendor Response Requirement Describe your system’s ability to support inbound calling, call control services, messaging for agents, speech recognition, text-to-speech, TDD and CTI and integration with a customer self-service interaction application.

Cisco Response: Cisco Unified IP IVR is designed to enhance the efficiency of any organization by simplifying business integration, increasing flexibility, and providing efficiency gains in network hosting. These features reduce business costs and can dramatically improve customer satisfaction. Enabled by Cisco AVVID, and tightly integrated with Cisco Unified CallManager, the Cisco Unified IP IVR offers ease of installation, configuration and application hosting because it is constructed specifically to exploit the power of IP-based communications. The Cisco Unified IP IVR is written entirely in Java and designed and constructed by Cisco to facilitate concurrent multimedia communication processing. The Cisco Unified IP IVR architecture is open and extensible, and it allows customers to incorporate custom-developed Java classes, which enables independent developers to extend the Cisco Unified IP IVR solution to meet unique customer business needs.

Key Features and Benefits of Cisco Unified IP IVR

Increase customer satisfaction

• Facilitate self-service applications such as access to checking account information or user-directed call routing by processing user commands via touch-tone input or speech recognition technologies

• Enable customers calling a contact center to use voice commands to retrieve the information they require without ever speaking with an agent, or—if required—to quickly navigate their way to the correct department or agent that can help them

• Provide multilingual support for Cisco Unified IP IVR Server prompts,



for ASR, and TTS • Deliver pro-active notification via eNotification to users through E-Mail,

fax, pager, and short message service (SMS) (some of these notification services require the use of a service provider or fax server)

• Provide more comprehensive and effective customer service by quickly efficiently handling call traffic via self service or transfer to the correct agent the first time

Reduce operating and application development costs

• Allow simple transactional requests to be offloaded from agents and handled by the IP IVR

• Enable rapid development and deployment of IVR applications via Web-based service creation and scripting environment

• Allow HTTP requests to trigger application execution via Hypertext Transfer Protocol (HTTP) support

• Supports optional automatic speech recognition (ASR) via integration with Nuance ASR software

• Supports optional text to speech (TTS) via integration with Nuance Vocalizer software

• Enable easy test and debug of applications using built-in debugging tools

• Provides standard real-time and historical reports to efficiently manage contact center resources

• Support development of customized reports using a third-party reporting package to meet additional reporting requirements in the contact center

• Significantly reducing costly T1 equipment required to deploy legacy IVR or private branch exchange (PBX) integration

Reduce acquisition, installation, and maintenance costs

• Use Cisco point-and-click single-CD installation for a quick and easy installation by any trained Windows NT/2000 system administrator

• Perform administration from anywhere on your corporate WAN via a completely Web-based administration interface

• Access to customer information to support self service applications or intelligent call routing via support for Open Database Connectivity (ODBC), access to Microsoft Structured Query Language (SQL) server, Oracle, IBM DB2, and Sybase databases

• Provides enhanced scalability for ease of expansion as your contact center operation grows

Optimize corporate and contact center resources

• Performs "prompt-and-collect" functions to obtain user data such as passwords or account identification that it can then pass to contact center agents

• Extract and parse Web content and present this data to customers through a telephony interface, thus facilitating the re-use and delivery of Web-maintained information to a caller through a voice portal

• Supports Voice Extensible Markup Language (VXML), which allows you



to create voice portals and voice-enabling Web sites that use existing Web technologies such as HTTP, XML, and Web or applications servers

• Integrates with CiscoWorks network management software

Component Details

Cisco Unified IP IVR has five primary components:

• IP IVR Editor—A Windows GUI-based application-development environment in which users create applications and flows using a drag-and-drop interface

• IP IVR Engine—The run-time environment that executes Cisco Unified IP IVR flows

• Step libraries—Libraries of JavaBeans, which provide the programming constructs, called steps, for the Cisco Unified IP IVR flows

• Flow repository (LDAP directory)—A storage location for all flows and configuration data

• Reporting Application—a tool that provides access to real-time and historical reporting statistics

The Cisco Unified IP IVR feature set processes voice-over-IP (VoIP) streams routed to the Cisco Unified IP IVR by the Cisco Unified CallManager and thus requires no digital-signal-processor (DSP) cards in the Cisco Unified IP IVR itself. Therefore, to connect the Cisco Unified IP IVR, Cisco Unified CallManager, and the Public Switched Telephone Network (PSTN), only one time-division multiplexing (TDM) interface is required: the VoIP gateway interface to the PSTN. IP-originated calls require no TDM infrastructure; therefore, Internet- and intranet-generated VoIP calls can terminate directly on the Cisco Unified IP IVR.

Administration of the Cisco Unified IP IVR is completely Web based; thus administrators with appropriate access can start, stop, and update the IP IVR Engine from a standard Web browser. The Web-based application administration manages the details of interfacing the Cisco Unified IP IVR with Cisco Unified CallManager and access to the other required network resources.

The IP IVR Engine also supports a powerful "debug mode" that allows the designer to create breakpoints and to directly upload and execute the flow in the IP IVR Engine for testing purposes. When testing is complete, the designer uploads the completed application to the directory, from where it may be deployed to any or all production Cisco Unified IP IVR servers on the network.

8.3.1 Script Development Vendor Response Requirement Describe the design tools/environment for IVR script development, the method used to test applications and changes prior to putting them into production and the method of putting changes into production.

Cisco Response: Cisco complies. The Cisco Unified IP IVR Application Editor is a Windows GUI-based tool with which script designers create new



flows (on the right side of interface) or modify existing flows, such as the Cisco automated attendant flow, which is included with the IP IVR as a sample script. The Cisco Unified IP IVR Editor is a visual scripting tool that allows designers to drag and drop flow steps from a palette into the main design window. Cisco Unified IP IVR flows, referred to in legacy products as scripts, are composed of a series of steps (on the bottom left of the interface) that are represented graphically in the Cisco Unified IP IVR Editor. The flow is the actual script that the user saves and executes on the IP IVR Engine. Steps are simply blocks of logic that application developers assemble into flows with the Cisco Unified IP IVR Editor to create custom IVR solutions. Downloadable to client workstations from the Web-based Cisco Unified IP IVR administration page, the Cisco Unified IP IVR Editor is supported on Windows 95, 98, NT Version 4.0, 2000, and XP. Flow developers can create scripts “offline” and import them into the live production environment at any time.

8.4.0 Workforce Management System The proposed solution must provide forecasting and scheduling capabilities as an option. Vendor Response Requirement Describe your system’s workforce management capabilities.

Cisco Response: Cisco complies. Cisco Unified Contact Center Enterprise leverages best of breed workforce management applications provided via



integrated third party partners such as Blue Pumpkin. Cisco has a rich API for integration, utilizing CTI data for scheduling and forecasting.

8.5.0 Integrated Email Call Control The proposed solution must integrate customer email messages as an option. It is also desirable that agents be able to handle a mix of voice and email messages on a call-by-call basis, and that all incoming voice calls and emails be routed into the same agent queue(s). Vendor Response Requirement Decribe your system’s capability to integrate email contact center functions with your voice call center system. Include information about the hardware and software requirements for this application.

Cisco Response: Cisco E-Mail Manager Option is a comprehensive, enterprise-class solution for managing high volumes of customer inquiries submitted to company mailboxes—or E-Mail submitted through your company's Web site—that will help you meet this challenge. Based on customizable business rules, the Cisco E-Mail Manager Option accelerates the response process by automatically directing messages to the most qualified agent or support team, categorizing and prioritizing messages, suggesting relevant response templates, and, if desired, sending automated replies. A full-featured, browser-based interface provides your agents with the productivity tools, knowledge, and resources they need to provide fast, accurate, and personalized responses to your customers. Moreover, the Cisco E-Mail Manager Option delivers the queue management, reporting, and outbound marketing tools that managers need in order to ensure that desired service standards are met, gain valuable insight into customer needs, and generate new revenue opportunities. Features and Benefits Sophisticated business rules—Cisco E-Mail Manager business rules promote efficient and detailed processing of free-form E-Mail, structured Web-forms, and other asynchronous media such as fax and voicemail. Because of the hierarchical and graphical rules building environment, call center supervisors or managers can easily create a sophisticated rule environment to match their internal business processes. In addition, these rules can be built without having to rely on costly IT or development support. Cisco E-Mail Manager rules environment includes automated messaging processing, which allows messages to be assigned to agents or teams, message prioritization and annotation, and offers automatic template suggestion to agents, auto acknowledgement, response, and text-pager notification. Flexible productivity tools—Designed to improve productivity in the contact center, these tools enable the agent to efficiently and effectively respond to E-Mail inquiries. A full-featured Web-based interface provides access to intuitive tools for reading, claiming, and responding to messages in the various queues. Templates for suggested or standard E-Mail replies enable



agents to insert suggested or standard replies directly into the response field for quick, authoritative, and consistent response creation. Moreover, in order to extend E-Mail response outside of the E-Mail Manager system, the MailTrack features enables external subject matter experts the ability to receive and respond to messages via their existing E-Mail client while maintaining full message tracking inside of E-Mail Manager. Other features include sender and thread communication history, "push" "pull" and "pick" message assignment options, template creation, spell check and queue management tools. Extensive management tools—provide managers and supervisors the capability to help them better manage resources in a busy call center environment. Powerful filtering and bulk message-management tools enable supervisors to adjust queue load based on message volume, time in queue, sender, and other message variables. A built-in reporting tool allows managers to generate reports on message traffic, system, agent and group performance, as well as customer inquiry trends. Finally, real-time monitoring provides current information on agent activity and performance and personal and skill group queue status, while queue status screens enable supervisors to closely monitor message queues to ensure that service-level goals are met. Enterprise-ready design—Cisco E-Mail Manager Option is designed to provide an E-Mail management system that is easy to install and configure with the powerful tools and scalability for enterprise-wide implementations. More than 100 configurable user role settings, response options, and mail-access permissions enable the system to accommodate the disparate needs of a large user base within a single or enterprise-wide implementation. Cisco E-Mail Manager Option offers an open application programming interface (API) that enables customers or third parties to readily integrate E-Mail Manager with customer relationship management (CRM) packages and other external data sources.

8.6.0 Web Center VoiceCon anticipates that it will require integration of its call center with its web server system. The proposed solution must support customer-initiated contact through the Internet as an option.



Vendor Response Requirement Describe how your call center can be integrated with the VoiceCon website to allow agents to respond to customer callback requests via the website. Include in the discussion whether agents can collaborate in realtime with callers during an online website transaction.

Cisco Response: The Cisco Web Collaboration Option provides your organization with a tool to increase sales, facilitate new revenue-generation opportunities, and enhance customer satisfaction and loyalty. Powerful Web collaboration features enable your contact center agents to deliver immediate answers to customer questions backed by Web pages and other Web-based content. Agents can also help customers solve complex support issues via simultaneous voice-and-visual interaction. In addition to allowing your contact center agents to share Web pages with customers while conducting a voice or text chat conversation, Web collaboration allows contact center agents and customers to collaboratively complete online forms, share any Windows desktop application using nothing more than a Web browser, and conduct one-to-one interactions and one-to-many or many-to-many online seminars. Such features as multi-session chat capabilities also improve the efficiency and productivity of your service representatives while decreasing customer wait times. By facilitating effective, personalized assistance that greatly enhances the customer experience, Cisco Web Collaboration Option is an ideal solution for both sales and service-oriented contact centers. It can be deployed in a pure IP environment, or seamlessly integrated with your organization's existing TDM telephony infrastructure to provide automated, blended delivery of phone and Web-based inquiries. Features and Benefits Powerful Web collaboration capabilities—The features include: text chat, bi-directional Web page sharing, Follow-Me-Browsing, bi-directional FormShare, real-time application sharing, collaborative white boarding, and ScriptBuilder for creating agent scripts of frequently shared Web pages and chat text. Customer-centric business communication—The Cisco Web Collaboration Option is designed to provide an unobtrusive and flexible communications environment for both the customer and the agent. To facilitate this, the following features are available:

o A comprehensive, easy-to-use interface for both the agent and the customer- both caller and agent interfaces run in a standard Web browser.

o Secure Sockets Layer (SSL) enabled for secure chat and FormShare. o Hands-free operation for customers—representatives can control all

two-way interactions. o Real-time network diagnostics for performance monitoring and

participant notification.



o Firewall-friendly communication over standard HTTP and HTTPS protocols.

ACD integration—The Cisco Web Collaboration Option easily integrates with your existing TDM telephony infrastructure, providing automated, blended delivery of Web-originated help requests. Customers requesting help can be automatically connected to an agent via telephone and Web collaboration all through a single help request. The result is a blending of traditional inbound voice calls with Web-based customer contact such as Web collaboration and text chat. Dynamic Content Adapter—This optional feature allows the Cisco Web Collaboration Option to share Web content or pages that are personalized, dynamically generated, or require passwords or log-ins. The ability to collaborate on secure, password-protected, and personalized Web content, without having to exchange passwords or other sensitive data, gives agents the ability to assist customers at the exact moment that they require help. Comprehensive management, reporting, and customization tools—These tools enable the Cisco Web Collaboration Option to be managed and customized to meet the requirements of the contact center. They can be used to build multi-user collaborative Java applications, create scripts that allow agents easy access to information that is frequently shared with customers—even split the customer's screen for side-by-side comparisons. Web-based administration, reporting, and management tools allow contact centers to easily produce comprehensive session and user data logging and management reports with drill-down capability, provide secure browser-based interfaces for all system management, and complete configurability of caller capabilities and experience.

8.7.0 Outbound Dialing The proposed solution must support automated outbound predictive dialing as an option. Vendor Response Requirement Describe your system’s capabilties to perform outbound predictive dialing, and include necessary hardware/software requirements.

Cisco Response: Cisco ICM Enterprise and Cisco Unified Contact Center Enterprise Editions enable companies to distribute inbound service volume to a variety of termination points, including automatic call distributors (ACDs), interactive-voice-response systems (IVRs), home agents, and network terminations. The addition of Cisco Outbound Option with its combination of outbound dialing modes complements the powerful inbound call-handling capabilities of the ICM and IPCC platforms with a robust outbound call management solution. The ability for agents to handle both inbound and outbound contacts offers a way to optimize contact center resources. The Cisco Outbound Option enables the multi-functional contact center to take advantage of the ICM enterprise management—allowing contact center managers in need of outbound



campaign solutions to take advantage of the enterprise view that Cisco ICM Enterprise and Cisco Unified Contact Center Enterprise maintain over agent resources. Features and Benefits Dialer—The Cisco Outbound Option dialer component is responsible for requesting campaign lists from the Outbound Option Campaign Manager, placing outbound calls, reserving agents, and classifying calls. Each dialer receives skill group information from the contact center solution. For IPCC, IP Dialer places outbound calls through a Voice Gateway via the Cisco Unified CallManager. For a traditional ACD (Avaya G3 and Avaya Multivantage), Outbound Dialer places outbound calls using Dialogic cards via the ACD. Dialing modes—Cisco Outbound Option offers three different dialing modes; predictive, preview, and progressive "power dialing."

Predictive—This mode is a dedicated dialing mode that handles contacts by continually adjusting the number of call originations needed per agent to achieve near 100-percent agent utilization. Preview—This mode enables the dialer to first reserve or "lock" an agent by placing a phantom call on the ACD. When reserved, Outbound Option posts a screen pop to the agent's CTI desktop, allowing the agent to dial, skip, or cancel the request. Progressive power dialing—This mode enables campaign administrators to configure the dialing rate such that Outbound Option does not attempt to predict the number of lines needed per agent and instead dials a set number of lines per agent. This number can be configured and set to a single line per agent. This feature attempts to guarantee that an agent is available when a customer answers his or her phone.

Blended modes—This feature enables Cisco ICM Enterprise and Cisco IPCC Enterprise to deliver both inbound and outbound (preview) calls to the same group of agents dynamically. Differing from the dedicated modes, blending enables an agent to switch between receipt of inbound and outbound calls on a call-by-call basis. This mode also employs the CTI desktop to deliver contact information to the agent either prior to an outbound call or as an inbound call is received. Campaign management—This feature enables call center managers to effectively administer and manage dialing campaigns by providing the ability to configure campaigns, create query rules, and import dialing lists. Campaigns can be configured to provide callbacks, which can be initiated based on the request of a party to be called back, or be automatically initiated by the system based on failure to contact a live party. Personal callback—This feature provides an agent the ability to insert a record into the Dialing List table to be called at a later time. The agent can then enter, directly into the desktop application, the time and date information most convenient for the customer to be contacted. The contact



record is modified in the Dialing List table, indicating a telephone number to call and a time to place the call, thereby allowing Cisco Outbound Option to offer the call to the specified agent at the designated time. Do Not Call list—This feature provides the capability to add entries into a Do Not Call table. By eliminating calls that are in the Do Not Call list, companies using the Cisco Outbound Option achieve a higher call completion success rate, while eliminating nuisance calls. To satisfy regulatory requirements, all outbound dialing solutions must support functionality to restrict dialing to specific contacts based on Do Not Call lists and permitted dial times. The Cisco Outbound Option enables managers to implement both of these functions based on regulatory requirements. Import Mechanism—Cisco Outbound Option employs a new import mechanism whereby imports can be kicked off as soon as an import text file is created (file polling). A third party application or batch process can create or copy an import text file as soon as it has contact information available. Imports can be kicked off at any interval the customer requires. Integration with the Cisco CTI Option—The Cisco CTI Option supports agents in blended mode by gathering call variables (for example, dialing list ID, time zone, account number, phone number, campaign ID and name, Customer Name and query rule ID) necessary to correctly handle the call, sending it to the appropriate agent or skill group and delivering customer information to the agent either prior to an outbound call or as an inbound call is received. Reporting—The sophisticated data storage design of the Cisco ICM Enterprise and Cisco Unified Contact Center Enterprise allow management of inbound and outbound calls within the same environment, enabling comprehensive reporting across the enterprise.

8.8.0 Server-Based CTI Call Control The proposed solution must support server-based CTI applications as an option. Vendor Response Requirement Describe the capabilities of the proposed solution to simultaneously route a call and data screen populated with the caller's identity, location or reason for calling.

Cisco Response: Cisco ICM Enterprise and Cisco Unified Contact Center Enterprise enable companies to deploy a complete network-to-desktop CTI strategy, including comprehensive functionality at the agent's workstation. These contact center solutions deliver a uniquely rich set of data to business applications, providing enterprise-wide call-event and customer-profile information to a targeted agent's desktop. This comprehensive CTI strategy enables the full utilization of corporate data in support of business rules and objectives at the point of customer contact. Cisco ICM Enterprise and Cisco Unified Contact Center Enterprise enable companies to more effectively utilize their corporate resources by taking



advantage of information collected from the Internet, carrier networks, automatic call distributors (ACDs), interactive voice response (IVR) systems, Web servers, E-Mail servers, databases, and other applications. Cisco ICM Enterprise and Cisco Unified Contact Center Enterprise set a new standard for true enterprise CTI by unifying contact center systems from the network to the desktop—and across the enterprise—with minimal custom development or systems integration, enabling an organization to implement CTI quickly and cost-effectively. Features and Benefits Advanced computer telephony connections that deliver rich CTI data—Cisco CTI Option provides a uniquely rich data set to the desktop through advanced computer telephony connections among the system and carrier networks, ACDs, IVRs, and Web and E-Mail servers. Information from these disparate resources is normalized by the ICM and IPCC system and then used in triggering integrated desktop telephony applications such as screen pops and voice and data transfers. Data elements can include dialed number, calling line ID (CLID), and customer-entered digits (CED); information submitted on a Web form; and information extracted from databases or created as a result of contact routing, voice processing, and agent transactions. Cisco ICM Enterprise and Unified Contact Center Enterprise route call-event and call-detail information to a targeted desktop when and where the contact is delivered. Screen pop—One of the most important aspects of a CTI implementation is the ability to provide caller data to the agent's application at the desktop. This is commonly referred to as a screen pop. To achieve the screen pop, information about the call and the caller is gathered from the network and provided to the CTI application. This information is provided along with information from the IVR to the desktop application, which looks up the customer's information and delivers the appropriate customer record screen to the agent's PC. A screen pop enables contact center agents to handle calls more quickly (agents are not required to query the customer for basic information), and provide better service (customers are not required to restate information that they may have already provided to an IVR). The screen pop information is also available to any agent who may subsequently join the call, for example, via a transfer or conference operation. Pre-route indications—Database lookups, particularly data-mining operations in which multiple records are queried, are time-consuming. The system delivers pre-route indications to a CTI client, enabling these database operations to take place before a contact arrives at a contact center and permitting complex data-mining results to be delivered to the desktop before contact arrival. When the system returns a route destination to the network, it simultaneously passes the customer-profile data to the CTI client. While the contact is still in the network, the client delivers appropriate database information to the agent desktop. As a result, screen pops occur when they are most useful—on or before the first ring of the agent's phone.





Part 2: System Pricing 1.0 System Pricing Requirements Summary system and voice terminal pricing data will be presented to VoiceCon workshop attendees and be deemed for public use. Detailed pricing data will remain confidential, and used to verify if the proposed system configurations satisfy RFP requirements. Installation fee pricing data is required, and must be included in the RFP response. Indicate if the proposed installation fee is based on direct sales/service or a channel partner pricing schedule. The proposed system price must also include a 1-year warranty to the customer. If this is a pricing option in your pricing schedule include it as part of the installation fee, and identify it as such.

Cisco Response: Pricing has been included in the tables below for the four required configurations. For budgetary purposes, installation costs have been estimated at 20% of list price; the exact installation costs will vary, depending upon the Cisco Partner involved in the design and installation of the system. A conservative discount of 25% has also been applied for this response (with no discount applied to the installation). This discount should not be construed as a guaranteed discount, and pricing is subject to change.



2.0 Summary Pricing – VOICECON NETWORK (all five locations) Complete and submit the attached EXCEL data table for your proposed system pricing summary data. The submitted data will be made available to the general public.

System Summary Pricing List

Price Discounted

Price All Common Equipment (call processing, port interfaces, media gateways, housings, power, feature/application servers, et al) $184,356 $138,267 Generic Software (Standard Features) (Including Device Licenses) $325,924 $244,443 Optional Software Features/Packages (Cisco Emergency Responder) IP Port License Fees (if applicable) $34,294 $25,721 Desktop Voice Terminals $611,010 $458,258 Systems Management/Administration System $66,604 $49,953 (Optional Operations Manager Package) Messaging System $216,696 $162,522 Installation Fee (including 1-year warranty) $300,596 $300,596 TOTAL $1,739,480 $1,379,759



3.0 Desktop Voice Terminal Pricing Complete and submit the attached EXCEL data table for your proposed Desktop Voice Terminal pricing summary data. The submitted data will be made available to the general public.

Voice Terminals

Model Quoted

List Price

Discounted Price (for this RFP)

Economy Desktop IP Telephone Instrument


$275 ($175 plus $100 Device License) $192.50

Administrative Desktop IP Telephone Instrument



Professional Desktop IP Telephone Instrument



Executive Desktop IP Telephone Instrument



IP Audio conferencing Unit

Cisco Unified IP Conference Station 7936


PC Client Softphone (Station User) License Fee

Cisco IP Communicator


PC Client Softphone (Attendant) License Fee No charge

Key Module Add-on Not quoted Gigabit Ethernet Module Add-on N/A* Display Module Add-on N/A WLAN Module Add-on N/A Desktop Power Module Option

Cisco IP Phone Power Cube (plus cord) $55 $41.25



*Note: Cisco offers three Unified IP Phone models with built-in Gigabit Ethernet switches: the 7941G-GE, 7976G-GE and 7971G-GE. 4.0 Detailed Configuration Components and Pricing Submit a separate EXCEL file with a detailed listing of proposed communications system components/elements and associated unit pricing, also indicating the proposed unit quantities included in the configuration for the base system (HQ facility). Also include an additional section with the configuration hardware/software elements and associated pricing data to satisfy each of the remote facilities (small, medium, large). Provide English language descriptions of all price configuration system components and elements in addition to any proprietary order codes. At minimum, the configuration component list should contain:

- All common control elements - All common equipment port cabinets/carriers - All port circuit interface cards for station and trunk ports - All media gateway equipment for station and trunk ports - All call control signaling interface cards - All voice terminals, including audioconferencing units - Generic software - All port license fees - All optional software packages

o Include all optional adjunct server equipment to support of required features o All voice messaging system elements (cabinet equipment and memory storage)

- All systems management elements The detailed pricing file will NOT be made public, but will be used by VoiceCon, only, to verify adherance to system configuration performance requirements and pricing summary data.

Cisco Response: Read and Understood. A separate pricing spreadsheet has been submitted with this proposal.



Appendix A:

CISCO UNIFIED CALLMANAGER VERSION 5.1

The Cisco® Unified Communications family of voice, video, and IP communications products and applications helps enable organizations to communicate more effectively-helping them streamline business processes, reach the right resource the first time, and reduce costs and maximize revenue. The Cisco Unified Communications system is an integral part of a complete, integrated business communications solution for organizations of all sizes that also includes network infrastructure, security, and network management products; wireless connectivity; a lifecycle services approach; and flexible deployment and outsourced management options, end-user and partner financing packages, and third-party communications applications.

Cisco Unified CallManager software is the call-processing component of the Cisco Unified Communications system. Cisco Unified CallManager extends enterprise telephony features and capabilities to packet telephony network devices such as IP phones, media processing devices, voice-over-IP (VoIP) gateways, and multimedia applications. Additional services such as unified messaging, multimedia conferencing, collaborative contact centers, and interactive multimedia response systems are made possible through Cisco Unified CallManager open telephony application programming interfaces (APIs). Cisco Unified CallManager is installed on the Cisco MCS 7800 Series of server platforms and selected third-party servers. It has a suite of integrated voice applications and utilities, including the Cisco Unified CallManager Attendant Console, an impromptu conferencing application, the Cisco Unified CallManager Bulk Administration Tool, the Cisco Unified CallManager CDR Analysis and Reporting Tool, the Cisco Unified CallManager Real-Time Monitoring Tool, and the Cisco Unified CallManager Assistant application.

FEATURES AND BENEFITS Cisco Unified CallManager 5.1 is an enterprise IP telephony call-processing solution that is scalable, distributable, and highly available. Multiple Cisco Unified CallManager servers are clustered and managed as a single entity on an IP network, a distinctive capability in the industry that yields scalability of 1 to 30,000 IP phones per cluster, load balancing, and call-processing service redundancy. Interlinking multiple clusters allows system capacity to reach 1 million users in a system of more than 100 sites. Clustering aggregates the power of multiple distributed Cisco Unified CallManager installations, enhancing the accessibility of the servers to phones, gateways, and applications, and triple call-processing server redundancy improves overall system availability. Call Admission Control (CAC) helps ensure that voice quality of service (QoS) is maintained across constricted WAN links, and it automatically diverts calls to alternate public-switched-telephone-network (PSTN) routes when WAN bandwidth is not available. A Web interface to the configuration database enables remote device and system configuration. HTML-based online help is available for users and administrators. Cisco Unified CallManager 5.1 builds upon the feature set available today with Cisco Unified CallManager 5.0 and 4.1(3). The appliance model provides a platform for call processing with the software preloaded on a Cisco Media Convergence Server (MCS)



platform; the software is optionally available as a DVD kit for customer-provided servers. The appliance comes with a single firmware image that includes the underlying operating system as well as the Cisco Unified CallManager application. The appliance is accessed through a GUI, and a command-line interface has been added to enable diagnostics along with basic systems management such as starting or stopping services and rebooting the appliance. No access to the underlying operating system is necessary. All systems management activities, such as disk space monitoring, system monitoring, and upgrades, are either automated or are controlled through the GUI. Because onboard agents are no longer supported on the appliance in this version, all Cisco Unified CallManager management interfaces are enhanced to allow for tight integration with third-party applications. The Simple Network Management Protocol (SNMP) interface has added an Overall Syslog performance MIB, the Serviceability interface has instrumented appliance-specific counters, and the Programming interface has added the ability to execute insert, update, and delete database commands. To further enhance security, Cisco Security Agent for Cisco Unified CallManager comes preloaded on the appliance. A host-based firewall has been added, along with IP Security (IPsec) connectivity between all cluster members. Session Initiation Protocol (SIP) support is available in Cisco Unified CallManager 5 with support of line-side devices, including IETF RFC 3261-compliant devices available from Cisco Systems® and other manufacturers. Cisco SIP-compliant devices include the Cisco Unified IP Phone 7905G, 7912G, 7940G, and 7960G models. SIP is also available on the Cisco Unified IP Phone 7911G, 7941G, 7941G-GE, 7961G, 7961G-GE, 7970G, and 7971G-GE models. The SIP trunk interface is available and conforms to RFC 3261, allowing support of video calls over the SIP trunk and improving conferencing and application support experiences when used with the Cisco Unity® and Cisco Unified MeetingPlace® solutions. Cisco Unified CallManager 5 supports Resource Reservation Protocol (RSVP) agent capability. The RSVP agent on a Cisco router extends CAC capability beyond a hub-and-spoke topology within a cluster. Now a call can be routed directly between two locations without having to traverse the hub, allowing alternative network topologies and more efficient use of networks. Cisco Unified CallManager 5.0 includes Japanese, Korean, and Chinese (Traditional and Simplified) languages, Cisco Unified CallManager 5.1 now supports Arabic too. SNMP is available to manage Cisco Unified CallManager, allowing managers to set and report traps on conditions that could affect service and send them to the remote monitoring systems. Cisco Unified CallManager allows immediate diversion of an incoming call or an in-progress call to voicemail. In Cisco Unified CallManager 5.1, the feature has been enhanced to address transferred calls as well. Users now have the option to forward calls that have been transferred to them either to their own voicemail or to the voicemail of the original transferring party. Cisco Unified CallManager provides a choice of operating system, either a Windows-based server (Release 4) or the appliance model (Release 5). The feature enhancements listed in this section are available only on the appliance model at this



time.

SPECIFICATIONS

Platforms • Cisco MCS 7800 Series, including Cisco MCS 7815, MCS 7825, MCS 7835, and MCS 7845

• Selected third-party servers; for details, visit: http://www.cisco.com/go/swonly

Bundled Software • Cisco Unified CallManager Version 5.1-Call-processing and call-control application

• Cisco Unified CallManager Version 5.1 configuration database-Contains system and device configuration information, including dial plan

• Cisco Unified CallManager administration software

• Cisco Unified CallManager CDR Analysis and Reporting Tool-Provides reports for calls based on call detail records (CDRs) that include calls on a user basis, calls through gateways, simplified call quality, and a CDR search mechanism. The Cisco Unified CallManager CDR Analysis and Reporting Tool also provides limited database administration; for example, deleting records based on database size.

• Cisco Unified CallManager Bulk Administration Tool (BAT)-Allows administrators to perform bulk add, delete, and update operations for devices and users

• Cisco Unified CallManager Attendant Console-Allows a receptionist to answer, transfer, and dispatch calls within an organization. The attendant can install the attendant console, a client-server application, on a PC running Windows 2000 or Windows XP. The attendant console connects to the Cisco Telephony Call Dispatcher (TCD) server for login services, line state, and directory services. Multiple attendant consoles can connect to a single Cisco TCD server.

• Cisco Unified CallManager Real-Time Monitoring Tool (RTMT)-A client tool that monitors real-time behavior of the components in a Cisco Unified CallManager cluster. Cisco Unified CallManager RTMT uses HTTP and TCP to monitor device status, system performance, device discovery, and computer-telephony-integration (CTI) applications. It also provides trace and log file management capabilities, including scheduling downloads of all trace and log files, user-defined events in trace and log files, and real-time monitoring of trace and log files. Cisco Unified CallManager RTMT can send E-Mail and page alerts when problems are detected. It connects directly to Cisco Unified CallManager by using HTTP for troubleshooting system problems.

• Cisco Conference Bridge-Provides software conference bridge resources for Cisco Unified CallManager

• Cisco Unified IP Phone Address Book Synchronizer-Allows users to synchronize Microsoft Outlook or Outlook Express address books with Cisco Personal Address Book. After installing and configuring Cisco Personal Address Book, users can access this feature from the Cisco Unified IP Phone Configuration Website.

• Cisco Unified CallManager Locale Installer-Provides user and network locales for Cisco Unified CallManager, adding support for languages other than English. The locales installer allows users to view translated text, receive country-specific phone tones, and receive Tool for Auto-Registered Phones Support (TAPS) prompts in a chosen language when working with supported interfaces. This application is downloaded from the Cisco Website as needed.

• Cisco Unified CallManager JTAPI-This plug-in is installed on all computers



hosting applications that interact with Cisco Unified CallManager with the Java Telephony API (JTAPI); JTAPI reference documentation and sample code are included

• Cisco Unified CallManager Telephony Service Provider-Contains the Cisco Telephony API (TAPI) service provider (TSP) and the Cisco Wave Drivers that enable TAPI applications to make and receive calls on the Cisco IP Telephony system

• Cisco Dialed Number Analyzer-Serviceability tool that analyzes the dialing plan for specific numbers

• Cisco Unified CallManager Assistant-Provides administration features along with administration Webpages for improved call handling

System Capabilities Summary Items marked with an asterisk (*) are new or enhanced for Cisco Unified CallManager 5.1.

• Alternate automatic routing (AAR)

• Attenuation and gain adjustment per device (phone and gateway)

• Automated bandwidth selection

• Auto route selection (ARS)

• AXL Simple Object Access Protocol (SOAP) API with performance and real-time information

• Basic Rate Interface (BRI) endpoint support; registers BRI endpoints as Skinny Client Control Protocol (SCCP) devices

• CAC-Intercluster and intracluster

• Call coverage

–Forwarding based on internal and external calls

–Forwarding out of a coverage path

–Timer for maximum time in coverage path

–Time of day

• Call display restrictions

• *Codec support for automated bandwidth selection: G.711 (mu-law and a-law), G.722, G.722.1, G.723.1, G.728, G.729A/B, GSM-EFR, GSM-FR, wideband audio (proprietary 16-bit resolution; 16-kHz sampled audio), and Advance Audio Coding (AAC) for use with Cisco Telepresence devices

• Digit analysis and call treatment (digit string insertion, deletion, stripping, dial access codes, and digit string translation)

• Distributed call processing

–Deployment of devices and applications across an IP network

–Virtual clusters of up to eight Cisco Unified CallManager servers for scalability, redundancy, and load balancing

–Maximum of 7500 IP phones per Cisco Unified CallManager server and 30,000 per server cluster (configuration-dependent)

–Maximum of 100,000 busy-hour call completions (BHCCs) per Cisco Unified CallManager server and 250,000 per server cluster (configuration-dependent)

–Intercluster scalability to more than 100 sites or clusters through H.323 gatekeeper



–Intracluster feature and management transparency

• Fax over IP-G.711 pass-through and Cisco Fax Relay

• Forced authorization codes and client matter codes (account codes)

• H.323 interface to selected devices

• H.323 FastStart (inbound and outbound)

• Hotline and private line automated ringdown (PLAR)

• Hunt groups-Broadcast, circular, longest idle, and linear

• Interface to H.323 gatekeeper for scalability, CAC, and redundancy

• *Divert calls to voicemail (iDivert)

• Language support for client-user interfaces (languages specified separately)

• Multilevel precedence and preemption (MLPP)

• Multilocation-Dial-plan partition

• Multiple ISDN protocol support

• Multiple remote Cisco Unified CallManager platform administration and debug utilities:

–Prepackaged alerts, monitor views, and historical reports with RTMT

–Real-time and historical application performance monitoring through operating system tools and SNMP

–Monitored data collection service

–Remote terminal service for off-net system monitoring and alerting

–Real-time event monitoring and presentation to common syslog

–Trace setting and collection utility

–Browse to onboard device statistics

–Clusterwide trace setting tool

–Trace collection tool

• Multisite (cross-WAN) capability with intersite CAC

• Dial-plan partitioning

• Off-premises extension (OPX)

• Outbound call blocking

• Out-of-band dual tone multifrequency (DTMF) signaling over IP

• PSTN failover on route nonavailability-AAR

• Q.SIG:

–Alerting name specified in ISO 13868 as part of the SS-CONP feature

–Basic call

–ID services

–General functional procedures

–Call back-ISO/IEC 13870: 2nd ed., 2001-07 (completion of calls to busy subscriber [CCBS] and call completion on no reply [CCNR])

–Call diversion, including SS-CFB (busy), SS-CFNR (no answer), and SS-CFU (unconditional); service ISO/IEC 13872 and ISO/IEC 13873, first edition 1995-Call diversion by forward switching and by reroute

–Call transfer by join

–H.323 Annex M.1 (Q.SIG over H.323)-ITU recommendation for Annex M.1



–Identification restriction: Calling Name Identification Restriction (CNIR), Connected Line Identification Restriction (COLR), and Connected Name Identification Restriction (CONR)

–Loop prevention, diversion counter and reason, loop detection, diverted to number, diverting number, original called name and number, original diversion reason, and redirecting name

–Message waiting indicator (MWI)

–Path replacement ISO/IEC 13863 2nd ed. (1998) and ISO/IEC 13974 2nd ed. (1999)

• Call preservation-redundancy and automated failover-on call-processing failure

• Station to station

• Station through trunk (Media Gateway Control Protocol [MGCP] gateways)

–JTAPI and TAPI applications enabled with automated failover and automatic update

–Triple Cisco Unified CallManager redundancy per device (phones, gateway, and applications) with automated failover and recovery

–Trunk groups

–MGCP BRI support (ETSI BRI basic-net3 user side only)

• Security

–Configurable operation modes-Nonsecure or secure modes can be configured.

–Device authentication-New model phones have an embedded X.509v3 certificate; a certificate authority proxy function (CAPF) is used to install locally significant certificate in the phones.

–Data integrity-The Transport Layer Security (TLS) cipher NULL-SHA is supported; messages are appended with the SHA1 hash of the message to ensure that they are not altered on the wire and can be trusted.

–Cisco Unified CallManager 5 offers secure HTTP support for Cisco Unified CallManager Admin, Cisco Unified CallManager Serviceability, Cisco Unified CallManager User, Cisco Unified CallManager RTMT, Cisco Unified CallManager Trace Analysis, Cisco Unified CallManager Service, Cisco Unified CallManager Trace Collection Tool, and Cisco Unified CallManager CDR Analysis and Reporting Tool.

–Privacy-Signaling and media are encrypted; including Cisco Unified IP Phone 7911G, 7940G, 7941G, 7941G-GE, 7960G, 7961G, 7961G-GE, 7970G, and 7971G models; Cisco Unified Survivable Remote Site Telephony (SRST); and Media Gateway Control Protocol (MGCP) gateways.

–Secure Sockets Layer (SSL) for directory-Supported applications include Cisco Unified CallManager BAT, Cisco Unified CallManager CDR Analysis and Reporting Tool, Cisco Unified CallManager Admin User Pages, Cisco Unified CallManager Assistant Admin Pages, Cisco Unified CallManager User Pages and Cisco Unified IP Phone Options Pages, Cisco Conference Connection, Cisco CTI Manager, Cisco CallManager Extension Mobility, and Cisco IP Manager Assistant.

–A USB eToken containing a Cisco rooted X.509v3 certificate is used to generate a Certificate Trust List (CTL) file for the phones and configure the security mode of the cluster.

–Phone security-Trivial File Transfer Protocol (TFTP) files (configuration and firmware loads) are signed with the self-signed certificate of the TFTP server;



the Cisco Unified CallManager system administrator can disable HTTP and Telnet on IP phones.

• SIP trunk (RFC 3261) and line side (RFC 3261-based devices)

• Cisco SRST

• Shared resource and application management and configuration:

–Transcoder resource

–Conference bridge resource

–Topological association of shared resource devices (conference bridge, music-on-hold sources, and transcoders)

–Media termination point (MTP)-Support for SIP trunk and RFC 2833

–Annunciator

• Silence suppression and voice activity detection

• Simplified North American Numbering Plan (NANP) and non-NANP support

• T.38 fax support (H.323 and SIP)

• Third-party applications support:

–Broadcast paging-Through foreign exchange station (FXS)

–Simplified Message Desk Interface (SMDI) for MWI

–Hook-flash feature support on selected FXS gateways

–TSP 2.1 interface

–JTAPI 2.0 service provider interface

–Billing and call statistics

–Configuration database API (Cisco AXL)

• Time-of-day, day-of-week, and day-of-year routing and restrictions

• Toll restriction-Dial-plan partition

• Toll-fraud prevention:

–Prevent trunk-to-trunk transfer

–Drop conference call when originator hangs up

–Require forced authorization codes

• Unified device and system configuration

• Unified dial plan

• Video codecs: H.261, H.263, H.264, and Cisco Wideband Video Codec (Cisco Unified Video Advantage)

• Video Telephony (SCCP, H.323, and SIP)

Summary of User Features Asterisks (*) in this list indicate SIP support for Cisco Unified CallManager 5.

• *Abbreviated dial

• *Answer and answer release

• *Autoanswer and intercom

• *Barge

• *Call-back busy, no reply to station

• *Call connection

• *Call coverage

• *Call forward-All (off net and on net), busy, and no answer



• *Call hold and retrieve

• Call join

• *Call park and pickup

• *Call pickup group-Universal

• *Call status per line (state, duration, and number)

• *Call waiting and retrieve (with configurable audible alerting)

• *Calling line identification (CLID) and calling party name identification (CNID)

• Calling line identification restriction (CLIR) call by call

• *Conference barge

• *Conference list and drop any party (impromptu conference)

• *Direct inward dial (DID) and direct outward dial (DOD)

• *Directory dial from phone-Corporate and personal

• *Directories-Missed, placed, and received calls list stored on selected IP phones

• *Distinctive ring for on- and off-net status, per-line appearance, and per phone

• *Drop last conference party (impromptu conferences)

• *Extension mobility support

• *Hands-free, full-duplex speakerphone

• *HTML help access from phone

• *Immediate divert to voicemail

• *Last number redial (on and off net)

• Malicious-call ID and trace

• Manager-assistant service (Cisco Unified CallManager Assistant application) proxy line support:

–Manager features-Immediate divert or transfer, do not disturb, divert all calls, call intercept, call filtering on CLID, intercom, and speed dials

–Assistant features-Intercom, immediate divert or transfer, divert all calls, and manager call handling through assistant console application

• Manager-assistant service (Cisco Unified CallManager Assistant application) shared-line support:

–Manager features-Immediate divert or transfer, do not disturb, intercom, speed dials, barge, direct transfer, and join

–Assistant features-Handle calls for managers; view manager status and calls; create speed dials for frequently used numbers; search for people in directory; handle calls on their own lines; immediate divert or transfer, intercom, barge, privacy, multiple calls per line, direct transfer, and join; send DTMF digits from console; and MWI status of manager phone

• Manager-assistant service (Cisco Unified CallManager Assistant application) system capabilities-Multiple managers per assistant (up to 33 lines) and redundant service

• Manager-assistant service on Cisco IP phones with Cisco Unified CallManager 5.1

• *MWI

• *Multiparty conference-Impromptu with add-on, meet-me features

• *Multiple calls per line appearance

• *Multiple line appearances per phone



• *Music on hold

• *Mute capability from speakerphone and handset

• *On-hook dialing

• Operator attendant-Cisco Unified CallManager Attendant Console: call queuing, broadcast hunting, and shared line support

• *Privacy

• *Real-time QoS statistics through HTTP browser to phone

• *Recent dial list-Calls to phone, calls from phone, autodial, and edit dial

• *Service URL-single-button access to IP phone service

• *Single directory number and multiple phones-Bridged line appearances

• *Speed dial-Multiple speed dials per phone

• *Station volume controls (audio and ringer)

• *Transfer-Blind, consultative, and direct transfer of two parties on a line

• *User-configured speed dial and call forward through Web access

• *Video (SCCP, H.323, and SIP)

• *Web services access from phone

• *Web dialer-Click to dial

• *Wideband audio codec support-Proprietary 16-bit resolution, 16-kHz sampling rate codec

Summary of Administrative Features • Application discovery and registration to SNMP manager

• AXL SOAP API with performance and real-time information

• Cisco Unified CallManager BAT

• CDRs

• Cisco Unified CallManager CDR Analysis and Reporting Tool

• Call forward reason code delivery

• Centralized, replicated configuration database and distributed Web-based management viewers

• Configurable and default ringer WAV files per phone

• Configurable call forward display

• Database automated change notification

• Date and time display format configurable per phone

• Debug information to common syslog file

• Device addition through wizards

• Device-downloadable feature upgrades-Phones, hardware transcoder resource, hardware conference bridge resource, and VoIP gateway resource

• Device groups and pools for large system management

• Device mapping tool-IP address to MAC address

• Dynamic Host Configuration Protocol (DHCP) block IP assignment-Phones and gateways

• Dialed Number Analyzer (DNA)

• Dialed number translation table (inbound and outbound translation)

• Dialed number identification service (DNIS)



• Enhanced 911 service

• H.323-compliant interface to H.323 clients, gateways, and gatekeepers

• JTAPI 2.0 computer telephony interface

• Lightweight Directory Access Protocol (LDAP) Version 3 directory interface to selected vendors' LDAP directories: Active Directory and Netscape Directory Server

• MGCP signaling and control to selected Cisco VoIP gateways

• Native supplementary services support to Cisco H.323 gateways

• Paperless phone DNIS-Display-directed button labels on phones

• Performance-monitoring SNMP statistics from applications to SNMP manager or to operating system performance monitor

• QoS statistics recorded per call

• Redirected DNIS (RDNIS) inbound and outbound (to H.323 devices)

• Select specified line appearance to ring

• Select specified phone to ring

• Single CDR per cluster

• Single point system and device configuration

• Sortable component inventory list by device, user, or line

• System event reporting to common syslog or operating system event viewer

• TAPI 2.1 CTI

• Time zone configurable per phone

• Cisco Unity software user integration

• TAPS

• Extensible Markup Language (XML) API into IP phones (Cisco Unified IP Phone 7940G and 7960G models)

• Zero-cost automated phone moves

• Zero-cost phone adds

• Data migration assistant

• Log partition monitor

• Disaster recovery framework

• Cisco Security Agent for Cisco Unified CallManager

• IPsec and certificate management

• CDR delivery manager

• Command-line interface

• Enhanced remote access through serial, console, and Secure Shell (SSH) Protocol

• Scheduled provisioning with Cisco Unified CallManager BAT

• Scheduled trace collection

• User-defined events

• Real-time trace monitoring

• Enhanced upgrade process to minimize service downtime

• Enhanced installation process to minimize install times

• Installation answer file for no-touch installs



• Syslog to SNMP trap MIB

• Enhanced AXL SOAP API to modify the database

CISCO UNIFIED CALLMANAGER VERSION 5 ENHANCEMENTS

SIP Trunk and Endpoint Support SIP trunk and endpoint support provides enhancements to support SIP and host SIP phones, improving interoperability and opening ways to develop innovative applications. Cisco Unified CallManager supports coexistence of SCCP and SIP phones, allowing migration to SIP while protecting investments in existing devices. Cisco Unified CallManager 5 includes the following major SIP functions:

• Native support of SIP devices

• CTI for Internet service provider (ISP) phones

• Presence information for SIP devices

• Fault, configuration, accounting, performance, and security (FCAPS) enhancements to support SIP

• SIP trunk enhancements for external applications, such as conferencing and presence

• Third-party SIP devices supporting RFC 3261

• SIP line-side RFCs: RFCs 3261, 3262, 3264, 3265, 3311, 3515, and 3842

• SIP trunk RFC support: RFCs 2833, 2976, 3261, 3262, 3264, 3265, 3311, 3515, 3842, 3856, and 3891

Licensing Application and phone software licenses are enforced. The system manages the maximum number of devices that can be provisioned.

• Each device (Cisco Unified IP phones, third-party devices, and video devices) provisioned in the system corresponds to a number of device license units (DLUs), depending on its capabilities; the total number of units is managed in Cisco Unified CallManager to determine capacity.

• DLUs must be purchased to cover the number of devices connected to Cisco Unified CallManager.

• Third-party SIP devices require DLUs for operation with Cisco Unified CallManager.

Localization The following user locales (languages) are supported: French, German, Italian, Spanish, Danish, Portuguese, Swedish, Norwegian, Dutch, Russian, Greek, Hungarian, Polish, Simplified Chinese, Traditional Chinese, Korean, Japanese, Brazilian Portuguese, Catalan, Croatian, Bulgarian, Slovak, Czechoslovakian, Slovenian, Romanian, Serbian, and now Arabic. Localization for Japanese, Korean, Traditional and Simplified Chinese, and Arabic are available with the Cisco Unified IP Phone 7911G, 7941G, 7941G-GE, 7961G, 7961G-GE, 7970G, and 7971G models. The following network locales (tones and cadences) are supported: Argentina, Australia, Austria, Belgium, Brazil, Canada, China, Colombia, Cyprus, Czech Republic, Denmark, Egypt, Finland, France, Germany, Ghana, Greece, Hong Kong, Hungary,



Iceland, India, Indonesia, Ireland, Israel, Italy, Japan, Jordan, Kenya, Korea Republic, Lebanon, Luxembourg, Malaysia, Mexico, Nepal, Netherlands, New Zealand, Nigeria, Norway, Pakistan, Panama, Peru, Philippines, Poland, Portugal, Russian Federation, Saudi Arabia, Singapore, Slovakia, Slovenia, South Africa, Spain, Sweden, Switzerland, Taiwan, Thailand, Turkey, United Kingdom, United States, Venezuela, and Zimbabwe.

Documents

Cisco Systems Unified Communications · PDF fileCisco Systems Unified Communications Proposal ... VoiceCon workshop attendees may use this RFP as a template for ... Cisco Systems,