Cisco Unified Communications System Proposal

Cisco Response To VoiceCon2009 RFP

Copyright © 2008 TEQConsult Group 1 of 264

Cisco Unified Communications

System Proposal

Based On:

Request for Proposal for an IP Telephony System

Prepared by Allan Sulkin

President, TEQConsult Group www.teqconsult.com

VoiceCon Orlando 2009



Preface

The following RFP document was exclusively designed and developed by TEQConsult for the VoiceCon ® Orlando 2009 Conference. The RFP is intended to solicit product information and pricing data about IP Telephony systems during the Winter 2008-2009 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 will be of benefit to 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, but only with expressed written permission from Allan Sulkin, TEQConsult Group ([email protected]). Proper accreditation to TEQConsult Group for using this RFP in whole or part must be included in the resulting document. Please note that this RFP is protected by US copyright laws and improper use by any means may result in legal action. TEQConsult Group would like to thank Fred Knight, GM, VoiceCon, for his review and editing of this document; Unimax Systems Corporation for its contributions to the systems management section of the RFP; and SecureLogix Corporation for its contributions to the security 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 where otherwise noted. 1.0.0 System Overview VoiceCon Company (VoiceCon) plans to install a new IP Telephony System (IPTS) at its newly constructed Campus (CHQ) facility. The IPTS will also support three remote facilities: Regional Office (RO); a Branch Office (BO); and Satellite Office (SO). Redundant IPTS common controllers must be installed in two equipment rooms distributed across the CHQ facility. All proposed common controllers must independently support all generic software features for the proposed IPTS as required in Section 5 of this RFP. The RO, BO and SO facilities will have local trunk services and be provisioned with survivable local common control elements behind the centrally located IPTS at CHQ. It is mandatory that a single system image IPTS solution be proposed that satisfies the requirements identified in this section of the RFP. The proposed IPTS must support 2,000 stations users distributed across the CHQ, RO, BO, and SO facilities at time of system cutover. Anticipated VoiceCon expansion plans will require the proposed IPTS be capable of supporting 50% growth of all call processing and port capacity parameters distributed proportionately across the four facilities. The proposed IPTS must support this growth requirement without replacement of any installed hardware or generic software (excluding updated releases). See Figure 1 for the VoiceCon IPTS network configuration:



CHQ

RO375 Stations

SO100 Stations

WAN

Voicecon IPTS Network

CHQ: Campus HeadquartersRO: Regional OfficeBO: Branch OfficeSO: Satellite Office

EquipmentRoom 1

750 Stations

EquipmentRoom 2

750 Stations

Figure 1

BO25 Stations

VoiceCon plans to install a 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 to support voice communications quality acceptable to VoiceCon. Pertinent bandwidth, latency, packet loss, and echo issues will be addressed by the LAN/WAN 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. NOTE: The proposed IPTS will be required to support a limited number of non-IP stations, e.g., analog telephones, requiring a RJ-11 connector. The proposed system may use circuit switched port carriers, LAN-connected media gateways, or some combination to support IPTS desktop analog communications and PSTN connectivity requirements. Vendor Response Requirement



Based on the RFP requirements prepare a simple network diagram that illustrates the proposed IPTS design. Include, at minimum, in the diagram the brand name/model of the IPTS, all common equipment (servers, circuit switched port carriers, media gateway equipment, systems management common equipment, required peripheral application servers, and messaging system equipment. Label all equipment components. Embed the diagram here and submit an original MS PowerPoint slide in a separate file. Please no copy/paste graphic from a different format, e.g. PDF.

Cisco Response: Cisco Unified Communications solutions unify voice, video, data, and mobile applications on fixed and mobile networks, delivering an easy-to-use, media-rich collaboration experience across business, government agency, and institutional workspaces. These applications use the network as the platform to enhance comparative advantage by accelerating decision time and reducing transaction time. The security, resilience, and scalability of the network enable users in any workspace to connect, everywhere, every time, everyone’s connected. Cisco Unified Communications is part of a comprehensive solution that includes network infrastructure, security, wireless, management, lifecycle services, flexible deployment and outsourced management options, and third-party applications.

Detailed information concerning Cisco Systems Voice and Unified Communications products and services can be found at:

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

Recently, Cisco Systems launched Cisco Unified Communications Release 7.0. This release introduced Cisco Unified Communications Manager 7.0 which delivers more than 40 new features built around three themes:

Reduced cost of ownership An open system that allows unified communications components to

interoperate transparently Improved user experience

Reduced Cost of Ownership - Cisco Unified Communications Manager 7.0 can help organizations lower total cost of ownership. Some of the new features that address cost of ownership include:

Local Route Groups and Transformation Patterns - reduce the configuration efforts involved in creating dial plans

Intelligent Bridge Selection - saves resources by optimizing the use of video bridge resources

Trusted Relay Points or Network Virtualization feature - facilitates trusted quality of service (QoS) and Call Admission Control (CAC), as well as trusted VLAN traversal for Cisco Unified Communications software clients



An Open System - An open systems approach to developing Cisco Unified Communications Manager enables the system to take advantage of a broader set of features from the Cisco Unified Communications solution components and other third-party products. Highlights of enhancements in this area include:

Expanded support of Session Initiation Protocol Ability for MobileConnect (single number reach) to support URLs as a

remote destination to a Microsoft Office Client Click-to-conference capability with Cisco Unified Presence and IBM

Sametime

Improved User Experience - As user needs evolve, Cisco Unified Communications Manager continues to meet those needs. Examples include:

Calling Party Normalization – simplifying global dialing plans and routing configurations

Cisco Unified Mobility: time-of-day settings and ability to enable/disable MobileConnect from an Unified IP Phone or Mobile Communicator

Dial via Office with Unified Mobile Communicator

Detailed information about Cisco Unified Communications Manager 7.0 can be found at:

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

The proposed Cisco IP Unified Communications solution for VoiceCon is shown in the diagram below. The details of the configuration are fully described in Sections 1.2.0 and 1.3.1 but the following are some highlights:

Cisco has proposed a complete Unified Communications system, including IP telephony, IP hard phones and soft clients, unified messaging, collaboration, presence and mobility applications. Unified Communications features are enabled for all users on the system.

Cisco Unified Communications Manager is installed in the two

Headquarters locations to provide a single system across all sites. One server is installed at each Headquarters site to support all the users and applications, and the two locations can back each other up as failover sites for all users. In addition, Cisco Unified Communications Manager is installed on servers located at the Regional Office, Branch Office and Satellite locations to provide fallback call processing for each of those sites.

Unified Communications applications including Unity Unified Messaging,

Cisco Unified MeetingPlace Express, Cisco Unified Presence Cisco Unified Mobility Manager and Cisco Unified Mobility Advantage are installed at the Headquarters site.

Cisco Emergency Responder for automated E911 support is included and

configured on redundant servers located at the two headquarters sites.



Cisco Integrated Services Routers (ISR’s) are located at each site to

provide PSTN Gateway services such as, FXS, FXO and T1 ports. VG248 Analog Gateways are installed at the two Headquarters and the Regional office site to provide FXS ports.

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 is requires manufacturer-specific LAN/WAN communications equipment to support any or all of the following voice communications operations or functions: call processing, port interface(s), network switching and routing, PoE, media gateway(s), QoS and security. Identify make/model of manufacturer-specific equipment if required.

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 Commercial Availability and Customer References The proposed IPTS equipment should be in current production and operating as part of a commercial system at a minimum of five (5) different customer installations.

Cisco Response: Read and Understood. Cisco complies. As of the date of submission of this RFP response* Cisco has over 100,000 customers worldwide, has shipped more than 20 million Unified IP Phones, and installed over 12 million Unity seats, 270 thousand MeetingPlace licenses, and 1.5 million IP Contact Center seats. Cisco has more than 400 customers who have deployed systems with more than 5,000 IP phones. In addition, Cisco has shipped more that 50 million VoIP gateway ports, and more than 130 million PoE-enabled Ethernet switch ports. *Data current as of 12 December, 2008

Vendor Response Requirement Confirm 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. NOTE: All proposed system hardware and software must be formally announced as of VoiceCon Orlando 2009 to be accepted by VoiceCon as a response to this RFP. This is a mandatory requirement. Indicate any unannounced capabilities at time of submission within answer to individual RFP requirement clause.

Cisco Response: Read and Understood. Cisco complies. All hardware and software versions referenced in this document will have been announced as of VoiceCon Spring 2009.

1.1.1 Single System Image



The proposed IPTS must provide to all system subscribers and administrators a Single System Image across VoiceCon CHQ, RO, BO and SO facilities. The Single System Image must include, but not be limited to, the following: 1) 4- or 5-digit dialing between all station users; 2) Virtually 100% transparent operation across all VoiceCon facilities and buildings of all station, attendant, and system features as identified in Section 5: Call Processing Features; 3) Headqiarter-housed systems management solution utilizing 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 capability to support a centrally located attendant pool; 5) Headquarter-housed messaging system resources; 6) Automatic alternate routing across the network for all voice calls (station-to-station and PSTN trunk connections). Vendor Response Requirement: Simply answer each of the following questions: 1. Is the proposed IPTS solution a true single system solution or multiple systems intelligently networked? 2. Does the proposed IPTS network solution fully satisfy all six (6) of the above Single System Image requirements? If not, identify and explain which of the requirements are not fully satisfied.

Cisco Response: The proposed Cisco Unified Communications Manager 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 Communications Manager cluster divided across the two headquarters locations. The Cisco Unified Communications system uses “clustering over the WAN” to provide full redundancy across these two locations. Servers can back each other up across the IP link between the two headquarters sites, and if that link is lost each site can operate independently. 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 office, branch, and satellite office. The proposed system fully satisfies all six of the stated single system image requirements. All supported features and applications such as, Presence, Mobility, Unity Unified Messaging, MeetingPlace, etc., 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 Communications Manager cluster.

1.1.2 Enhanced 911 (E911) Services Support It is mandatory that the proposed IPTS support E911 services provided by a



public safety answering point (PSAP) as defined by FCC regulations. Each of the four (4) VoiceCon facilities utilizes a different PSAP. All VoiceCon IPTS station user E911 calls must be directed to their local PSAP for call handling and response regardless of location. Only identify and discuss the E911 solution that is included in the pricing proposal.

Cisco Response: Cisco Response: Cisco Unified Communications Manager 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 Communications Manager 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 Communications Manager 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.

Vendor Response Requirement: Confirm that the proposed communications system solution supports E911 service for all user stations (IP and analog) regardless of physical location. Also, briefly explain how E911 service requirements are supported, specifically addressing each of the following issues/questions:

1) A brief description of any optional hardware/software equipment, including peripheral servers. [Note: Include the price of all required equipment, including servers, in the pricing proposal]

Cisco Response: Cisco Emergency Responder has been included in this proposal and is included in the total system price under the “Optional Software” category. The Cisco Emergency Responder solution consists of two MCS 7816-



H3 servers (redundant) and the server and user licences to support 2000 users. This application is very cost effective. At the discounted price quoted for this proposal, it costs $18,597 which includes redundant servers. (See pricing summary under “Optional Software”). This adds about $9 per line to support VoiceCon’s 2,000 user configuration, and provides a fully automated E9-1-1 solution that provides near real-time updates of location information. 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 Communications Manager, 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 Communications Manager, 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.

Cisco Emergency Responder proactively queries Cisco Unified Communications Manager 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.

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.

2) Identify any redundant hardware/software elements included in the solution.

Cisco Response: Cisco has included a second Emergency Responder server (MCS 7816-H3) to form a fully redundant Cisco Emergency Responder Group with the same capacity and increased availability compared with a single Cisco Emergency Responder server.

3) Are all E911 calls routed to the local PSAP from each of the four (4)

VoiceCon facilities?

Cisco Response: Yes. 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 Communications Manager 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.

4) How are E911 calls placed from a mobile communications device, e.g.

802.11 wireless handset, handled?

Cisco Response: Device mobility brings about special design considerations for emergency calls. Cisco Emergency Responder (Cisco ER) can be used to track device mobility and to adapt the system's routing of emergency calls based on a device's dynamic physical location, typically based on the 802.11 access point location. In a centralized call processing deployment, Cisco ER cannot fully support device movement across call admission control locations because Unified CM does not know about device movements. For example, if you physically move a phone from Branch A to Branch B but the phone's call admission control location remains the same (for example, Location_A), then it is possible that calls made to 911 from that phone would be blocked due to call admission control denial if all available bandwidth to Location_A is in use for other calls. This call blocking occurs even if the phone, now in location B, is physically co-located with the gateway used to connect to the PSAP for location B. For the same reasons, Cisco ER cannot support device movement across gatekeeper-controlled call admission control zones. However, Cisco ER can fully support device movements within a call admission control location. In centralized call processing deployments, Cisco ER automatically supports device movement within branches. However, if a device is moved between branches, manual intervention is required to adapt the device's location and region parameters before Cisco ER can fully support 911 calls. In cases where soft clients such as Cisco IP Communicator are used within the enterprise, Cisco ER can provide device mobility support. However, if the soft client is used outside the boundaries of the enterprise (for example, VPN access from a home office or hotel), Cisco ER will not be able to determine the location of the caller. Furthermore, it is unlikely that the Cisco system would have a gateway properly situated to allow sending the call to the appropriate PSAP for the caller's location. It is a matter of enterprise policy to allow or not to allow the use of soft clients for 911 calls. It is highly advisable to disallow 911 calls by policy for soft clients that can roam across the internet. Nevertheless, if such a user were to call 911, the best-effort system response would be to route the call to either an on-site security force or a large PSAP close to the system's main site. The following



paragraph is an example notice that you could issue to users to warn them that emergency call functionality is not guaranteed to soft client users:

“Emergency calls should be placed from phones that are located at the site for which they areconfigured (for example, your office). A local safety authority might not answer an emergency call placed from a phone that has been removed from its configured site. If you must use this phone for emergency calls while away from your configured site, be prepared to provide the answering publicsafety authority with specific information regarding your location. Use a phone that is locally configured to the site (for example, your hotel phone or your home phone) for emergency calls when traveling or telecommuting.”

5) How are analog and IP desktop (telephone instrument and PC client soft

phone) station user moves/adds/changes reported to the PSAP?

Cisco Response: In general, you must associate a single fully qualified E.164 number, known as the emergency location identification number (ELIN), with each ERL. (However, if using Cisco Emergency Responder, you can configure more than one ELIN per ERL.) The ELIN is used to route the call across the E911 infrastructure and is used by the PSAP as the index into the ALI database. ELINs must meet the following requirements:

They must be routable across the E911 infrastructure. If an ELIN is not routable, 911 calls from the associated ERL will, at best, be handled according to the default routing programmed in the E911 selective router.

Once the ERL-to-ELIN mapping of an enterprise is defined, the corresponding ALI records must be established with the LEC so that the ANI and ALI database records serving the PSAP can be updated accurately.

Once the ERL-to-ELIN mapping is established, it needs be modified only when there are changes to the physical situation of the enterprise. If phones are simply added, moved, or deleted from the system, the ERL-to-ELIN mapping and its associated ANI/ALI database records need not be changed. CER tracks the internal location and will send the appropriate ELIN to the PSAP ensuring the accuracy of the location information.

6) What degree of specificity for calling station user location is identified to

the E911 PSAP? Desktop work area, local switch room, work floor, other?

Cisco Response: 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.

The National Emergency Number Association (NENA) has recently proposed model legislation to be used by state and federal agencies in enacting the rules that govern 911 in enterprise telephony systems. One of the concepts in the NENA proposal is that of the emergency response location (ERL), which is defined as:

A location to which a 911 emergency response team may be dispatched. The location should be specific enough to provide a reasonable opportunity for the emergency response team to quickly locate a caller anywhere within it.

Rather than having to identify each phone’s location individually, the requirement allows for the grouping of phones into a "zone," the ERL. The maximum size of the ERL may vary, depending upon local implementation of the legislation. For example, the legislation may specify a square footage such as 7000 square feet (sq ft), and that would form the basis for assigning ERLs. So, while it is possible to assign an ERL down to the level of each individual phone, it is more typical for customer to group phones into logical locations, for example, by building, floor, etc.

7) Can E911 calls be simultaneously routed to an internal security desk in addition to the PSAP?

Cisco Response: 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.

8) Are E911 calls logged and recorded separately from non-E911 calls for

reporting purposes?

Cisco Response: 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.

1.1.2.1 E911 and Station Moves



Station user moves behind the proposed IPTS solution must be tracked dynamically in real time for E911 services support. Vendor Response Requirement: Indicate if the proposed E911 solution satisfies this requirement and indicate how often the local and PSAP databases are updated.

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 IPTS Vendor Response Requirement Provide the following information regarding your proposed IPTS as details are requested in following sections:

1. Product and model name(s) for the IPTS and messaging system 2. Software release for each product/model proposed 5. Product/model commercial availability 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, etc. 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, Branch, and Satellite sites. The Cisco Unified Communications Manager servers at the two headquarters locations act as the primary centralized call controllers. Backup Cisco Unified Communications Manager servers are located in each remote site to provide local backup in the event of a WAN failure at that site.

Cisco Unified Workspace Licensing (CUWL) has been used in this proposal to provide VoiceCon with an easy and affordable program to procure a broad range of Cisco Unified Communications applications and services. Cisco Unified Workspace Licensing, inclusive of all client and server software, licensing, service and support, software subscription for applications and clients, facilitates consistent deployment of multiple applications to all users in their workspaces, and helps organizations maximize the potential of unified communications.



This program streamlines pricing, licensing, software subscription and acquisition of Cisco Unified Communications solutions and introduces the ability for businesses, government agencies, and institutions to implement a media-rich unified communications experience at a lower per-user price point. Presence, unified clients, mobility, unified messaging, and audio, video, Web conferencing are just a few of the applications included in this program. What is Cisco Unified Workspace Licensing?

Cisco Unified Workspace Licensing offers two options, Standard Edition and Professional Edition. CUWL is charged per user, and the one fee entitles the customer to server and client licenses and three years of upgrades. The following table compares the two options.

Note: The Professional Edition was used in this proposal for VoiceCon. Pricing details are discussed in Part 2 “System Pricing”.

Functionality What’s Included in the Workspace License

Standard Edition

Professional Edition

Video Conferencing

Cisco Unified MeetingPlace Express (25 Cisco UWL=1 Port)

N Y

Web Conferencing

Cisco Unified MeetingPlace/MeetingPlace Express Port (25 CUWL=1 Port)

N Y

Audio Conferencing


N Y

Mobile Phone Client

Cisco Unified Mobile Communicator Client

N Y

Contact Center

Cisco Unified Contact Center Express

N Y

Presence Cisco Unified Presence Profile

Y Y

Mobility (with Sim Ring services)

Cisco Unified Mobility Profile

Y Y

Soft Client Cisco Unified Personal Communicator or Cisco Unified IP Communicator with Cisco Unified Video Advantage

Y Y

Microsoft MOC Integration

Cisco Integration for Microsoft MOC

Y Y

Messaging Cisco Unity or Cisco Unity Connection

VM VM/UM

Phone/Call Control

License for One or Unlimited Cisco IP Phones per User

One Unlimited

The following Cisco equipment and applications have been included in this proposal to provide the complete solution (broken down by



Headquarters, Regional Office, Branch and Satellite locations): Headquarters Location: The primary headquarters location contains several Cisco Unified Communications components (these are the total quantities, which will be installed in the two locations at headquarters):

Call Processing and Application Servers Unified Communications Manager Release 7.0 on three (3) Cisco

Media Convergence Server (MCS) 7835-H2 appliances. These servers are redundant, load sharing call processing servers, configured as one Publisher, and two subscribers.

One (1) MCS 7845-H2 server for Cisco Unity Unified Messaging and Automated Attendant (96 ports).

One (1) MCS 7845-H2 server for Cisco MeetingPlace Two (2) MCS 7816-H3 servers for Cisco Emergency Responder One (1) MCS 7835-H2 server for Cisco Presence One (1) MCS 7835-H2 and one (1) MCS 7825-H2 server for

Cisco Unified Mobility Advantage Cisco Unified IP Phones

Cisco Unified IP Phone 7906G – 50 Cisco Unified IP Phone 7962G plus 7915 Expansion Module – 100 Cisco Unified IP Phone 7965G - 700 Cisco Unified IP Phone 7975G - 450 Cisco 7937 IP Conference Station – 22 SolutionsPlus ARC Enterprise PC Attendant Console - 6 Cisco Unified Personal Communicator (softphone) licenses (for

users at all locations)– 1743 Cisco Unified Mobile Communicator client licenses – 1743 (for

users at all locations) PSTN Gateways

Six (6) Cisco Integrated Service Routers, Model 2811. o Six (6) NM-HDV2-2FT1/E1 IP Communications High-Density

Digital Voice Network Module with 2 T1 ports providing 12 T1 ports

o Twenty-four (24) VIC2-4FXO modules for a total of 96 FXO ports.

Analog FXS Gateways Four (4) Voice Gateway (VG) 248 to provide 144 FXS ports for

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

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

Regional Office Location: The regional office location contains several Cisco Unified Communications components:

Call Processing Server (Local Backup) Unified Communications Manager Release 7.0 on one (1) Cisco

Media Convergence Server (MCS) 7825-H2 appliance. Cisco Unified IP Phones

Cisco Unified IP Phone 7906G – 15 Cisco Unified IP Phone 7962G plus 7915 Expansion Module – 15



Cisco Unified IP Phone 7965G - 210 Cisco Unified IP Phone 7975G - 100 Cisco 7937 IP Conference Station – 8 SolutionsPlus ARC Enterprise PC Attendant Console - 2

Cisco 2811 Integrated Services Routers Three (3) Cisco Integrated Service Routers, Model 2811.

o Two (2) NM-HDV2-2 T1/E1 – 2-port T1 module, providing 4 T1 ports

o Five (5) VIC2-4FXO 4-port voice interface cards for a total of 20 FXO ports

o Seven (7) VIC-4FXS/DID – 4-port FXS module, providing 28 FXS ports

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

Branch Location: The branch location contains the following Cisco Unified Communications components:



Cisco Unified IP Phone 7906G – 5 Cisco Unified IP Phone 7962G plus 7915 Expansion Module – 5 Cisco Unified IP Phone 7965G - 45 Cisco Unified IP Phone 7975G - 30 Cisco 7937 IP Conference Station – 5

Cisco 2811 Integrated Services Routers Two (2) Cisco Integrated Service Routers, Model 2811 Two (2) NM-HDV2-2 T1/E1 – 2-port T1 module, providing 2 T1

ports Three (3) VIC-4FXS/DID – 4-port FXS module, providing 12 FXS

ports Two (2) VIC2-4FXO, four-port FXO modules for a total of 8 FXO

ports Power Failure Transfer Unit


Satellite Location: The satellite location contains the following Cisco Unified Communications components:



Cisco Unified IP Phone 7906G – 5 Cisco Unified IP Phone 7962G plus 7915 Expansion Module – 3 Cisco Unified IP Phone 7965G - 10 Cisco Unified IP Phone 7975G - 5

Cisco 2811 Integrated Services Router Two (2) VIC-4FXS/DID module, four-port FXS modules for a total



of 8 FXS ports Two (2) VIC2-4FXO, four-port FXO modules for a total of 8 FXO

ports Power Failure Transfer Unit


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

Converged TDM/IP: call telephony server(s) supporting LAN/WAN distributed circuit switched port interface cabinets with integrated media gateway interfaces for non-IP/IP port (station and trunk) connectivity

Soft switch (client/server): call telephony server(s) supporting LAN-connected media gateway equipment for non-IP/IP port connectivity

For either design call telephony server to IP endpoint (station and trunk) call control signaling (including call setup/teardown, feature/function access and implementation) should be supported through a direct transmission connection over the LAN/WAN. Vendor Response Requirement: Briefly provide an overview description of the proposed IPTS solution architecture and design. Include in your response information that directly addresses the following:

1. Architecture design: converged or soft switch (as defined above) 2. Call telephony server(s) and associated required common control

equipment 3. Circuit switched port carrier/interface equipment, if applicable 4. LAN-connected media gateways (server-embedded, standalone,

switch/router-equipped, desktop), if applicable

Cisco Response: Cisco is proposing a distributed IP unified communications system based on Cisco’s IP Communications architecture. Call processing and applications servers are centralized (and divided across the two Headquarters locations). PSTN Gateways, IP and analog phones, and other resources are distributed across VoiceCon’s Headquarters, Regional office , branch and satellite locations.

1.3.1 Common Control There are several mandatory common control requirements. 1.3.1.1 Common Control Housing



VoiceCon requires that all IPTS common control elements be fully embedded in compact housing with internal interfaces to media gateways, non-IP port circuit interfaces, and service circuit boards, as applicable. Call control signaling to/from all IP endpoints must be supported through an integrated Ethernet LAN uplink connector, e.g., RJ-45. The cost benefits of this fully integrated design are reduced hardware, power, and system footprint requirements. Vendor Response Requirement: Confirm this requirement is satisfied.

Cisco Response: Cisco Media Convergence Servers provide highly available server platforms to host applications within the Cisco Unified Communications system. These platforms address enterprise customer requirements for Cisco Unified Communications Manager installations for up to 30,000 IP phones within a single Cisco Unified Communications Manager cluster. Cisco Media Convergence Servers also provide platforms for:

Cisco Emergency Responder

Cisco IP Interoperability and Collaboration System (IPICS)

Cisco Unified Contact Center Enterprise


Cisco Unified Contact Center Hosted

Cisco Unified Intelligent Contact Management Enterprise

Cisco Unified IP Interactive Voice Response

Cisco Unified IP Queue Manager

Cisco Unified MeetingPlace and MeetingPlace Express

Cisco Unified Mobility Advantage

Cisco Unified Presence

Cisco Unity messaging solutions

The MCS 7835-H2 Unified Communications Manager Appliance has been proposed for Cisco Unified Communications Manager. The MCS-7825-H3 has been proposed as backup servers in the remote sites. The MCS 7845-H2 has been proposed for Cisco Unity Unified Messaging. The MCS 7835-H2 has been proposed for Cisco Unified Presence and Cisco Unified Mobility Advantage. The MCS 7816-H3 has been proposed for Cisco Emergency Responder. Cisco Media Convergence servers are an integral part of a complete, scalable architecture for a new generation of high-quality IP voice solutions that run on enterprise data networks, and they deliver the high performance and availability demanded by today's enterprise networks. The Cisco Media Convergence servers are described in more detail below. MCS 7845-H2 At just 2 rack units (2RU) high, the Cisco MCS 7845-H2 offers tremendous power in a low profile chassis that minimizes rack space. The appliance



supports up to 7500 Cisco Unified IP phones per appliance and 30,000 Cisco Unified IP phones per cluster, and includes the following features and components:

• Two Intel Woodcrest Xeon 2.33-GHz processors, a 1333-MHz front side bus (FSB), and 4 MB of Layer 2 cache

• 4-GB PC2-5300 667-MHz double data rate 2 (DDR2) memory with online spare capabilities

• Smart Array P400 Redundant Array of Independent Disks (RAID) Controller with 256-MB cache

• Dual-port Gigabit Ethernet controller (embedded)

• Quick-deployment third-party rail kit

• Support for Integrated Lights Out 2 (iLO2) server management

• Support for up to eight small form-factor pluggable (SFP) hard drives

• Hot-plug redundant power supplies

• Hot-plug redundant fans MCS 7835-H2 The Cisco® MCS 7835-H2 is a high-availability server platform for Cisco Unified Communications solutions. The server appliance is preinstalled with an operating system and Cisco Unified Communications Manager 7.0 and supports up to 2,500 users. It is fully operational upon startup, requiring entry of just a few configuration items such as IP address and domain. At only 2 rack units (2RUs) high, the Cisco MCS 7835-H2 offers tremendous power in a low-profile chassis that minimizes rack space. Features of the Cisco MCS 7835-H2:

• Intel 5140 Xeon 2.33-GHz processors, an 1333-MHz front side bus (FSB), and 4 MB of Layer 2 cache

• 2-GB PC2-5300 667-MHz double-data-rate 2 (DDR2) memory with online spare capabilities




• Support for Integrated Lights Out 2 (iLO 2) server management

• Support for up to 8 small form-factor hot-plug hard drives

• Hot-plug redundant power supplies

• Hot-plug redundant fans

MCS 7825-H3 The Cisco MCS 7825-H3 Media Convergence Server is an entry-level server platform used for Cisco Unified Communications solutions. It can support up to 1,000 users. At only 1 rack unit (1RU) high, the Cisco MCS 7825-H3 offers tremendous power in a low-profile chassis that minimizes rack space and offers the following features:

• Intel Dual-Core Xeon 3050 2.13-GHz processor with an 1066-MHz front side bus (FSB) and 2 MB of Layer 2 cache



• 2-GB PC2-5300 double-data-rate 2 (DDR2) Error Checking and Correcting (ECC) memory

• Two 160-GB cold-swap serial advanced technology attachment (SATA) hard disk drives configured with Redundant Array of Independent Disks (RAID) 1



MCS 7816-H3 The Cisco® MCS 7816-H3 Media Convergence Server is an entry-level server platform used for Cisco Unified Communications solutions. At only 1 rack unit (1RU) high, the Cisco MCS 7816-H3 offers tremendous power in a low-profile chassis that minimizes rack space and offers the following features:

• Intel Celeron D 352 3.2-GHz processor with an 533-MHz front side bus (FSB) and 512 KB of Layer 2 cache

• 2-GB PC2-5300 double-data-rate 2 (DDR2) Error Checking and Correcting (ECC) memory

• 160-GB cold-swap serial advanced technology attachment (SATA) hard disk drive


• Quick-deployment third-party rail kit 1.3.1.1.1 Common Control Vendor Response Requirement: Briefly describe the hardware housing for all -located common control elements (call processing, signaling, et al), specifically including size (H x L x W), weight (standard common assembly), fan cooling units, and all embedded hardware components.

Cisco Response: The following table summarizes the hardware proposed at each site. This includes servers for call control and all other applications, as well as PSTN and analog gateways. Additional details are provided for each of the servers in Section 1.3.1.1 above.

Hardware Function Qty H x W x L

(Inches) Weight (lbs)

Fans

CHQ-1 MCS -7835 Unified Communications Manager

2 3.38 x 17.54 x 26.01 60 6

MCS -7845 Unity 1 3.38 x 17.54 x 26.01 60 6 MCS -7845 MeetingPlace 1 3.38 x 17.54 x 26.01 60 6 MCS -7835 Presence 1 3.38 x 17.54 x 26.01 60 6 MCS -7835 Mobility Advantage 1 3.38 x 17.54 x 26.01 60 6 MCS -7825 Mobility (Proxy server) 1 1.70 x 16.78 x 24.0 27 6 MCS -7816 Emergency Responder 1 1.70 x 16.78 x 24.0 27 4 2811 ISR PSTN Gateways 3 3.5 x 17.4 x 16.4 25 3 VG248 Analog Gateway 2 1.75 x 17.25 x 16.75 14 5 CHQ-2 MCS -7835 Unified Communications 1 3.38 x 17.54 x 26.01 60 6



Manager MCS -7816 Emergency Responder 1 1.70 x 16.78 x 24.0 27 4 2811 ISR PSTN Gateways 3 3.5 x 17.4 x 16.4 25 3 VG248 Analog Gateway 2 1.75 x 17.25 x 16.75 14 5 RO MCS -7825 Unified Communications

Manager 1 1.70 x 16.78 x 24.0 27 6

2811 ISR PSTN Gateways and SRST

3 3.5 x 17.4 x 16.4 25 3

Branch MCS -7825 Unified Communications

Manager 1 1.70 x 16.78 x 24.0 27 6


2 3.5 x 17.4 x 16.4 25 3

Satellite MCS -7825 Unified Communications

Manager 1 1.70 x 16.78 x 24.0 27 6


1 3.5 x 17.4 x 16.4 25 3

1.3.1.1.2 Remote Survivable Common Control Vendor Response Requirement: Briefly describe the hardware housing for all remotely- common control elements (call processing, signaling, et al) used for local survivability, specifically including size (H x L x W), weight (standard common assembly), fan cooling units, and all embedded hardware components.

Cisco Response: The Regional Office, Branch and Satellite Office locations are equipped with Media Convergence Servers, MCS 7825-H3, running Cisco Unified Communications Manager 7.0 for local bacup. These servers are included in the table above in Section 1.3.1.1.1

1.3.1.2 Common Control Redundancy The IPTS common control must be based on a fully redundant, i.e. duplicated, design. Redundant components may be provisioned as active/passive or load sharing with seamless switchover operation between the elements in case of errors or failure. All active calls and programmed feature states must be preserved during the switchover process. Vendor Response Requirement: Confirm that the proposed IPTS common control fully satisfies the requirement for a fully redundant, i.e. duplicated, design and identify if it is based on an active/passive or load sharing design by completing the following table.



Element

Active/Passive, Load Sharing, or Both (Indicate if Not Applicable)

Primary call processor

Load Sharing

Main system memory

Load Sharing

Customer database memory

Load Sharing

RJ-45 Ethernet uplinks

Active/Passive

Power supply

Load Sharing

Tone generators

Not applicable

Call classifiers

Not applicable

Registers

Not applicable

DTMF receivers

Not applicable

I/O interfaces

Not applicable



Cisco Response: The highly simplified diagram above illustrates the redundant Cisco Unified Communications Manager (CUCM) configuration proposed for VoiceCon. At the headquarters locations, three (3) Cisco Unified Communication Manager Appliances, model MCS 7835-H2, are included in the configuration as load sharing, redundant Unified Communications Manager call processing servers. One is a Publisher and two are subscribers. These two subscribers are the main call processing servers for the system. In addition, one CUCM appliance, model MCS 7825-H3 is included at each remote location to act as a local backup server in the event that the phones at that site lose connectivity to the primary subscribers at the headquarters site. These six CUCM servers are configured as a single Communications Manager cluster. Unified Communications Manager software and all configuration databases are fully duplicated and mirrored on the six servers. These servers are rated at 15,000 BHCC’s per server (MCS 7835-H2) and 6,000 BCCS’s per server (MCS 7825-H2), and (60,000 BHCC’s per cluster). The cluster is divided across all locations, however the two main subscribers are located at headquarters (CHQ1 and CHQ2) to provide “clustering over the WAN” (or in this case, clustering split between two locations on the same headquarters campus network). Two servers are located at CHQ1, and one server is located at CHQ2 . This configuration ensures that each site can back up the other site across the network links. (Note: VoiceCon should provide separate redundant links between CHQ1 and CHQ2 to ensure the highest availability). If both network links fail between CHQ1 and CHQ2, each of these sites can operate independently. It should be noted that the servers are sized such that only one server is required to support all 2,000 users on the system. To grow to 3,000 users in the future, one additional MCS 7835-H2 server would need to be added to the cluster. Each device would be configured with three levels of redundancy by putting the device into a redundancy group according to its location. So, failover would follow the sequences shown below: Headquarters 1

Primary server CUCM-HQ1 (Subscriber) Secondary server CUCM-HQ2 (Subscriber) Tertiary server CUCM – HQ1 (Publisher)

Headquarters 2 Primary server CUCM-HQ2 (Subscriber) Secondary server CUCM-HQ1 (Subscriber) Tertiary server CUCM – HQ1 (Publisher)

Regional Office Primary server CUCM-HQ1 (Subscriber) Secondary server CUCM-HQ2 (Subscriber) Tertiary server CUCM – RO (Subscriber)

Branch Office Primary server CUCM-HQ1 (Subscriber) Secondary server CUCM-HQ2 (Subscriber) Tertiary server CUCM – BO (Subscriber)

Satellite Office



Primary server CUCM-HQ1 (Subscriber) Secondary server CUCM-HQ2 (Subscriber) Tertiary server CUCM – SO (Subscriber)

If a switchover is required between Unified Communications Manager servers, calls in progress are preserved. The customer configuration database is mirrored on the six 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.

The MCS 7835-H2 Unified Communications Manager 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 Communications Manager software and therefore are also fully duplicated. Cisco Unity Unified Messaging is configured on one MCS 7845-H2 server (non-redundant). MeetingPlace and Mobility Advantage are also confured on single servers. The Cisco Emergency Responder is configured as a redundant application on two servers located at CHQ1 and CHQ2.

1.3.1.3 Distributed Control The redundant common controllers must be distributed between the two equipment rooms, e.g. one (1) or more call telephony call servers in each equipment room based on system design architecture.

Vendor Response Requirement: Confirm that redundant common control elements are distributed and installed at each of the two Headquarter equipment rooms. Describe how the distributed common control elements are physically and logically linked across the two equipment rooms, including cabling requirements.

Cisco Response: Cisco complies. Cisco Unified Communications Manager redundancy is fully described in Section 1.3.1.2 above. The redundant servers are connected across the VoiceCon campus network and WAN.

1.3.1.4 Call Processing Vendor Response Requirement: Confirm that the proposed IPTS can handle a minimum of 100,000 Busy Hour Call Completions (BHCCS) in its proposed and priced system design and equipped configuration.



Cisco Response: The MCS 7835-H2 servers are rated at 15,000 BHCCs per server, and 60,000 BHCCs per cluster.

1.3.2 Call Processor Make/Model Vendor Response Requirement: Identify the make/model of the main processor element for the proposed common control system.

Cisco Response: The Cisco MCS 7835-H2 Unified Communications Manager includes an Intel 5140 Xeon 2.33-GHz processor, an 1333-MHz front side bus (FSB), and 4 MB of Layer 2 cache.

1.3.3 Call Processing O/S Vendor Response Requirement: Identify the primary operating system used by the main processor element of the proposed common control system. Linux is preferred, but not mandatory.

Cisco Response: Linux for Cisco Communications Manager 7.0. 1.3.4 Memory Vendor Response Requirement: Briefly describe system memory and storage design for both generic software and customer database requirements. Include in response storage capacities.

Cisco Response: The Cisco MCS 7835-H2 Unified Communications Manager includes:

• 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 2000 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 800 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: Does the proposed IPTS solution come 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 (not included in the pricing for this proposal) 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 all call telephony servers (or equivalent hardware) the number of available and configurable RJ-45 Ethernet LAN uplink interfaces for call control signaling to LAN-connected cabinets/carriers, media gateways, and IP ports. Include a brief description of how the physical Ethernet connection is provided: dedicated circuit board; daughterboard; fully embedded 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 the rear of the 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 Communications Manager, 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 Communications Manager. So in most customer deployments the servers and the gateways derive clock from NTP, and the phones derive clock from Unified Communications Manager. 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).

1.4 Remote Survivability In standard operating mode station users at remote facilities will be supported by Headquarters common control. It is required that station users at the remote facilities be provided with full, uninterrupted access to all IPTS features and services regardless of Headquarters common control failure or LAN/WAN connectivity problems due to switch, router, or private network transmission service errors or failures. Each remote facility must perform a seamless switchover to the local common control while all active calls (intercom and



trunk) calls and programmed feature states, e.g. call forwarding, are preserved. Vendor Response Requirement: Confirm that the proposed IPTS solution satisfies this requirement for remote local survivability and briefly describe the local hardware/software solution at each of the three remote facilities necessary to support the seamless switchover operation specifically addressing preservation of all call types and programmed feature states. Indicate the time it takes (in seconds) to perform the switchover if not instantaneous and if the customer can optionally program the switchover time (in seconds) to accommodate infrequent short disruptions in LAN/WAN transmission signaling. Also indicate if station users are warned via telephone/PC soft phone display of a delay in dial tone and call implementation occurring during the switchover process.

Cisco Response: At the headquarters locations, three (3) Cisco Unified Communication Manager Appliances, model MCS 7835-H2, are included in the configuration as load sharing, redundant Unified Communications Manager call processing servers. One is a Publisher and two are subscribers. These two subscribers are the main call processing servers for the system. In addition, one CUCM appliance, model MCS 7825-H3 is included at each remote location to act as a local backup server in the event that the phones at that site lose connectivity to the primary subscribers at the headquarters site. These six CUCM servers are configured as a single Communications Manager cluster. Unified Communications Manager software and all configuration databases are fully duplicated and mirrored on the six servers. Therefore, in fallback mode at



a remote site, all features are preserved. The cluster is divided across all locations, however the two main subscribers are both located at headquarters (CHQ1 and CHQ2) to provide clustering over the WAN (or in this case, clustering split between two locations on the same headquarters campus network). Two servers are located at CHQ1, and one server is located at CHQ2 . This configuration ensures that each site can back up the other site across the network links. (Note: VoiceCon should provide separate redundant links between CHQ1 and CHQ2 to ensure the highest availaibility). If both network links fail between CHQ1 and CHQ2, each of these sites can operate independently. It should be noted that the HQ1 and HQ2 servers are sized such that only one server is required to support all 2,000 users on the system. The MCS 7825-H3 servers located at each remote site can support 1,000 users and are intended for local backup, not systemwide backup. Each device would be configured with three levels of redundancy by putting the device into a redundancy group according to its location. So, failover would follow the sequences shown below: Headquarters 1

Primary server CUCM-HQ1 (Subscriber) Secondary server CUCM-HQ2 (Subscriber) Tertiary server CUCM – HQ1 (Publisher)

Headquarters 2 Primary server CUCM-HQ2 (Subscriber) Secondary server CUCM-HQ1 (Subscriber) Tertiary server CUCM – HQ1 (Publisher)

Regional Office Primary server CUCM-HQ1 (Subscriber) Secondary server CUCM-HQ2 (Subscriber) Tertiary server CUCM – RO (Subscriber)

Branch Office Primary server CUCM-HQ1 (Subscriber) Secondary server CUCM-HQ2 (Subscriber) Tertiary server CUCM – BO (Subscriber)

Satellite Office Primary server CUCM-HQ1 (Subscriber) Secondary server CUCM-HQ2 (Subscriber) Tertiary server CUCM – SO (Subscriber)

So, as an example, IP Phones at the Satellite Office would be supported by the centralized CUCM servers at HQ1/HQ2, but if they are unable to reach those servers for any reason, they would fallback to the local server (CUCM-SO) located in the Satellite Office.

If a switchover is required between Unified Communications Manager servers, calls in progress are preserved. The customer configuration database is mirrored on the six 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. Of course, in the event of a WAN failure, where connectivity is lost to the Headquarters site, access to centralized applications such as, Unity, Presence, etc., would be lost.



How long does it take? 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).

Unified Communications Manager call processing service shuts down gracefully such that it tears down the TCP connections with the IP phones.

Otherwise, the failover will take place after Cisco Unified Communication manager fails to acknowledge three consecutive kee-palive messages (sent at configurable interval) from the Cisco IP Phones.

Do users get a warning? Users on active calls will get a warning suggesting that the Unified Communication Manager is down and that none of the mid-call features are available on the active call owing to the fact that the switchover only takes place for that IP phone upon call termination.

1.4.1 Survivable IPTS Features/Services VoiceCon requires full feature/function survivability at the two largest remote facilities (Regional and Branch offices). Basic call features/functions, only, are acceptable at the smallest facility (Satellite Office). Vendor Response Requirements: Identify any required generic software feature (See Section 5.0 Call Processing Features) or system function (including E911 support services) not available or operational when the local survivability option is activated at either the RO or BO facility. As an alternative response provide a list of the survivable features/functions if a significant number are not supported. Also identify any type of station user equipment (telephone instruments, consoles, soft phones, mobile clients, et al) not supported in local survivability mode at any of the three remote facilities.

Cisco Response: Since the CUCM cluster is a single system image, the customer configuration database is mirrored on the six 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. Of course, in the event of a WAN failure, where connectivity is lost to the Headquarters site, access to centralized applications such as, Unity, Presence, Emergency Responder etc., would be lost. And, calls in progress on the WAN itself would be lost, but subsequent calls would use alternate routing to use the PSTN to bypass the WAN failure.

1.4.2 Local Survivability Failover/Switchback For each of the remote facilities operating in local survivability failover mode



switchback to the Headquarters common control should not result in interruption or disconnection of any in-process telephony services. Vendor Response Requirements: Describe the switchback process when the Headquarters common control is operational and accessible, specifying if the process is automatic or manual, identifying how long the process takes to implement. Positively confirm that connected calls and/or telephony operations at the remote facility are not affected in any way during the switchback process. Describe how calls and services are affected if the answer is negative.

Cisco Response: Upon restoration of the WAN connectivity, the system automatically shifts all call processing and telephony features back to the centralized Cisco Unified Communications Manager cluster at headquarters. During the switchback process, the connected calls are maintained and all the telephony operations (mid-call features) are available for end users. In terms of switchback process, the Cisco IP Phone periodically sends a test registration message to the primary Unified Communication Manager. If the primary server is available and can acknowledge the registration request, the IP phone unregisters with the secondary server that it is currently registered with and registers back to the primary server. If the primary server cannot handle the failback, it rejects the test registration request with a retry value.

1.4.3 Survivable Messaging Services It is desirable that remote station users at the three remote facilities have access to messaging services (see Section 7) if there is a WAN link disruption to the HQ messaging system. Vendor Response Requirements Briefly describe how messaging services would be accessed and implemented by remote station users when there is a WAN link disruption. Would access to and implementation of any messaging features/functions be affected in this situation?

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.

Alternatively, users with Cisco Unified Mobile Commmunicator on their smarphones can retrieve messages at any time from any location. (see Section for details on Cisco Unified Mobile Communicator).



1.4.4 Network Failover Resiliency VoiceCon desires that the proposed IPTS solution support network failover resiliency to a comparable IPTS solution located at a future VoiceCon facility in the unlikely event of a catastrophic Headquarters common control failure, i.e., all common control elements. Network failover resiliency requires that all IP endpoints, cabinets/port carriers, and media gateway at all facilities (Headquarters, Regional, Branch, Satellite) automatically re-register to a designated back-up IPTS if so programmed. Vendor Response Requirements Respond to each of the following:

Can the proposed IPTS solution support a network failover resiliency operation to a back-up IPTS in case of a catastrophic common control failure?

If yes, briefly describe the failover process including the time required before full telephony services are available to re-registered station users.

Can there be more than one designated back-up IPTS? If yes, how many?

Cisco Response: A Cisco Unified Communications Manager 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 Communications Manager cluster primary subscribers have been split across two sites: CHQ1 and CHQ2. 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 (or MAN). Using this deployment model, you may have up to eight sites with Cisco Unified Communications Manager subscribers being backed up by Cisco Unified Communications Manager 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.



Note: In the design proposed for VoiceCon, the two Heaquarters sites are on the same LAN, so failover between the two primary servers is not considered to be “clustering over the WAN” since there are no delay or bandwidth limitations between those two sites. The comments below apply if VoiceCon were to choose to truly separate their data centers across a WAN link, and are included for reference.

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 Communications Manager 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 Communications Manager 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 Communications Manager 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 Communications Manager Service enabled. The connections are a full mesh between these servers. Because only eight servers may have the Cisco Communications Manager 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 Communications Manager servers. This traffic is marked as priority ICCS traffic and exists between the Cisco Unified Communications Manager server with the CTI Manager and the Cisco Unified Communications Manager server with the CTI device.

Failover Between Subscriber Servers

With Cisco Unified Communications Manager Release 7.0, the device configuration records are cached during the initialization or boot-up time. The effect is that Cisco Unified Communications Manager 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 Design Standards



1.5.1 Session Initiated Protocol (SIP) VoiceCon requires that the proposed IPTS support SIP-compatible stations and trunk networking as specified by IETF Work Group RFC documents, including 3261, 3263, 3264, 3265, 3604 and 4456. Vendor Response Requirements Respond to each of the following:

Does the proposed IPTS conform to each of IETF Work Group RFC documents cited above? Also list any other IETF Work Group RFC documents supported by the proposed IPTS solution.

Is the proposed IPTS solution based on a native-SIP design or is optional hardware/software required?

Can the proposed IPTS support SIP-compliant desktop telephone instruments if required?

Can the proposed IPTS support SIP trunk services if required? Indicate if optional SIP proxy gateways are required.

Can the proposed IPTS support SIP-enabled applications, such as Internet conferencing, telephony services and features, presence, events notification and instant messaging? Indicate if optional server equipment is required for any of the listed applications.

Cisco Response: Cisco Unified Communications Manager Release 7.0 supports IETF standard SIP IP Phones. These include Cisco Unified IP Phones with SIP firmware loads as well as third-party SIP Phones. Cisco Unified Communications Manager supports RFC 3261 compliant devices, so as long as the third party phones are compliant they should work with Unified Communications Manager Release 6.1. (Appendix A shows a detailed list of the SIP RFC’s that are supported.) SIP call control is native to Unified Communications Manager 7.0 (i.e., embedded) and does not require additional hardware or software components. Note that Cisco Unified Communications Manager 7.0 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 Communications Manager and Cisco Unified Communications Manager 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 Communications Manager and Unified Communications Manager Express.

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

SIP Clients: The Cisco Unified IP Phones included in this proposal support IETF Standard SIP. They can be ordered as SCCP (Cisco Skinny protocol) phones and later field upgraded with SIP firmware loads if desired. The Cisco 7937 Conference Station and the Cisco Unified Personal Communicator included in this response are only available as SCCP endpoints at this time. The Cisco Unified Personal Communicator softphone will support SIP in a future release. The 7937 Conference Station however will not. SIP Trunking: Cisco has supported SIP-based trunking since Unified Communications Manager Release 4.0. SIP call control is native to Unified Communications Manager (i.e., embedded) and does not require additional hardware or software components. Note that Unified Communications Manager 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 Communications Manager 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.

SIP trunks provide connectivity to other SIP devices such as gateways, proxies, voicemail systems, and other Unified CM clusters. Cisco Unified CM 5.x and 6.x introduced major enhancements for SIP trunks and removed the limitations in Cisco Unified CM 4.x, such as single codec support, lack of video support, and the mandatory media termination point (MTP) for RFC 2833 DTMF support.

The main enhancements to SIP trunks in Cisco Unified CM 6.x are the support for the iLBC and AAC codecs and SIP PUBLISH. By providing improved performance, SIP PUBLISH provides the preferred mechanism for Cisco Unified CM 6.x to send IP phone presence information to Cisco Unified Presence over a SIP trunk.

Major enhancements to SIP trunks in Cisco Unified CM 7.x include:

Support for offering the G.729 codec in outgoing initial SIP INVITE requests

Usage of Privacy, P-Asserted-Identity, and P-Preferred-Identity headers to signal the content and whether to display or restrict calling and called party names and numbers



Encryption of media using Secure Real-Time Transport Protocol (SRTP)

Support for normalization of calling party numbers

For the complete list of new enhancements for SIP trunks, refer to the Cisco Unified Communications Manager product release notes available at

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

When used for intercluster trunking, SIP trunks do not support QSIG Tunneling using Annex M1.

1.5.2 Services Oriented Architecture (SOA) Does the proposed IPTS solution support SOA standards to facilitate design and development of new features/functions and/or interaction with business process solutions? If yes indicate if SOA supported is fully embedded in the system design or if optional hardware/software is required. Describe optional hardware/software if applicable.

Cisco Response: The Cisco Unified Communications system is the first unified communications system takes full advantage of converged networks and service-oriented architectures (SOA). Based on Cisco integrated technology, it uses shared services and open standards such as Extensible Markup Language (XML), Voice XML (VXML), Session Initiation Protocol (SIP), SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE), HTTP, and Simple Object Access Protocol (SOAP) to virtualize voice, video, presence, and mobility services within the Smart Business Communications framework. These services can be delivered securely to any device, anywhere, anytime, across multiple applications – in a plug-and-play format that dramatically improves efficiency and collaboration, and transforms the user experience. Customers and developers can leverage these standards to create custom applications that enhance the functionality of their Cisco Unified Communications system. Cisco Unified Application Environment Cisco also offers an application development environment to simplify this development process called Cisco Unified Application Environment (CUAE). CUAE includes the Cisco Unified Application Designer which is a visual integrated development environment (IDE) that facilitates the development of applications that converge voice and video with enterprise applications and data to transform business processes, improve communications, and create competitive advantage. As widespread adoption of IP communications moves responsibility for telephony into the corporate data center, developers in the IT department are increasingly assuming responsibility for the development of applications that must incorporate voice and video capabilities. Most IT developers have little to no experience with arcane telephony protocols, media processing, and the other technologies that accompany voice and video development. As a result, converged telephony and data application development projects require too



much time and money, yield functionally limited applications, and are prone to failure. The Cisco Unified Application Designer allows developers with no telephony development experience to rapidly, easily, and successfully build feature-rich converged voice, video, and data applications. Developers use drag-and-drop techniques to visually construct applications using simple communications business logic. Developers can focus on what they want to do rather than spending all their time trying to figure out how to do it. Using this approach, development time for new applications is dramatically reduced to days or weeks instead of months

A public beta of Cisco Unified Application Environment 2.5. Cisco Unified Application Environment 2.5 is available that delivers a number of new features and capabilities, such as:

Support for third-party integrated development environments, including Eclipse and Microsoft Visual Studio, and new language support, such as Java and C#. Support for Python, Ruby, and Javascript are on the roadmap.

Ability to run applications and plug-ins anywhere, not just on Cisco Unified Application Environment.

A new management architecture that includes a new java-based Web console with a redesigned user interface, and a programmatic API for automating administrative functions.

Cisco Unified Application Environment 2.5 is pre-release software and should not be used in production environments. All applications and plug-ins written for 2.4 will work with 2.5 without any changes.

Key Features and Benefits

Graphical Application Definition - The Cisco Unified Application Designer allows developers to visually construct applications by dragging and dropping prebuilt functions onto a graphical communications business logic canvas and visually updating parameters associated with the graphical functions. Developers can focus on what they want to do and rapidly assemble feature-rich applications without needing to learn and struggle with all the low-level details associated with the technologies used by the application.

Application Integrity Checks - The Cisco Unified Application Designer automatically checks the application being designed for common syntax and logic errors. If the Cisco Unified Application Designer finds any problems with the application, it notifies the developer so that the developer can proactively fix the problem.

Extensible Toolbox - The Cisco Unified Application Designer offers an extensible toolbox with built-in visual application functions for popular telephony call-control protocols such as Session Initiation Protocol (SIP), H.323, Skinny Client Control Protocol (SCCP), and Java Telephony Application Programming Interface (JTAPI), as well as functions for other IP communications protocols such as Cisco IP Phone Services, DeviceListX, AVVID XML Layer Simple Object Access Protocol (AXL-SOAP), Extension Mobility, and



other Cisco Unified Communications Manager (formerly known as Cisco Unified CallManager) APIs. In addition to common telephony protocols, the toolbox includes built-in functions for integrating with common enterprise applications and data, such as Web Services, HTTP, Lightweight Directory Access Protocol (LDAP), Structured Query Language (SQL), and Simple Mail Transfer Protocol (SMTP). The toolbox is completely extensible, so Cisco customers and partners can add support for any standards-based or proprietary protocol or other interface.

Instant Web Services Integration - Developers can easily make any Web service available for visual application construction. If a developer simply points to the Web Services Description Language (WSDL) for a Web service, the Cisco Unified Application Designer makes the Web service functions available in the toolbox for easy drag-and-drop access and integration.

Embedded Code - Developers can achieve a great deal simply by using the functions available from the toolbox to visually construct applications. If a developer needs something unique, the Cisco Unified Application Designer offers a .NET-compliant code editor with which developers can write custom code for unique functions required by an application.

Runtime Debugging - When an application is developed, the Cisco Unified Application Designer provides a built-in runtime debugger to help the developer rapidly find and fix errors. The debugger supports breakpoints set by the developer at any point within the communications business logic. The developer can also use a real-time break function to stop execution of the application at arbitrary points in time, or a single step and continue function to walk through the application one step at a time. While the application is running, the developer can easily monitor the state of application variables and examine the call stack.

One-Click Deployment - When satisfied with the quality of the application, the developer can deploy the application to the Cisco Unified Application Server with a single click, deploying the application for use across the organization's worldwide IP communications infrastructure in moments.

Product Specifications - The table below lists the specifications of the Cisco Unified Application Designer.

Product Specifications

Item Specification

Product compatibility

Compatible with Cisco Unified Communications Manager and Cisco Unified CallManager Versions 3.3, 4.0, 5.0, 6.0 and 7.0; Cisco Unified Communications Manager Express and Cisco Unified CallManager Express Versions 4.0 and 5.0; Cisco Unified Presence 1.0 and 6.0; Cisco Unified IP Phones; Cisco IP Communicator; and Cisco Unified Application Server 2.4 and Cisco Unified Media Engine 2.4



Protocols SIP, H.323, SCCP, JTAPI, Cisco IP Phone Services, DeviceListX, AXL-SOAP, Extension Mobility, Cisco Unified Communications Manager APIs, Web Services, HTTP, LDAP, SQL, and SMTP; easily extensible to any standards-based or proprietary protocol or interface

1.6 Security

VoiceCon requires a secure IPTS network solution to optimize system performance and reduce the probability of toll fraud, restricted calls, and illegal system and network access.

1.6.1 Unauthorized System Access VoiceCon requires that the proposed IPTS solution be secure against unauthorized system access. The following system design and configuration guidelines should be followed:

All unnecessary ports, such as telnet, SNMP, etc. will be closed by default.

The software running for ports will not contain any known vulnerabilities.

Administrative interfaces will not ship with known default passwords. Default community strings for SNMP will not be used. SNMP version 3

will be supported. The switch network will support security features such as VLANs,

Network Admission Control (NAC), and other features. Key components, such as the call processor, media gateway, or

associated servers/cards will have built in host-based intrusion prevention systems.

Vendor Response Requirements Confirm that the proposed IPTS solution satisfies each of the above listed security attributes. 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 Communications Manager is done via HTTPS, which uses server certificate authentication and client username/password authentication. Access to the Unified Communications Manager 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 Communications Manager 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. Communications Manager 5.0 introduced configuration file encryption using AES-128-CBC (Cipher-Block Chained). Beginning in Unified Communications Manager 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 Communications Manager 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 Communications Manager, 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 Communications Managers, 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 Communications Manager. This is done as part of the TLS establishment during the phone’s registration with Unified Communications Manager. In this way, mutual authentication occurs when phones register with Unified Communications Manager. 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-CBC 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 cipher-block chaining, designated as AES-128-CBC, for encryption of signaling. A lock icon next to a Unified Communications Manager’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 Communications Manager is both authenticated and encrypted. Finally, Unified Communications Manager provides a rich set of Toll fraud counter-measures. These capabilities are integral to the Unified Communications Managers 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 Unauthorized Network Access VoiceCon requires that the proposed IPTS solution be secure against unauthorized network access. Vendor Response Requirements Briefly describe call type detection and prevention processes embedded in the proposed IPTS solution to identify and prevent:

Unmonitored and non-secured Internet sessions by employees calling private Internet Service Providers accounts using modems connected to corporate phone lines.

Unlawful data network access by outsiders penetrating through modem-enabled corporate phone lines connected to LAN/WAN accessible workstations and other equipment.

Cisco Response: An analog line must be used for modems, which would require the configuration of the port on the gateway, the physical run of the cable, and configuration of Cisco Unified Communications Manager. Only if those criteria were met could a modem ever dial out of the corporate network. It’s possible to configure call routing (Partitions and Calling Search Spaces) within Cisco Unified Communications Manager to disallow anyone from calling that modem thus blocking access from outsiders dialing into the corporation.

1.6.3 Disruption of Services VoiceCon requires that the proposed IPTS solution be secure against disruption of services. A minimum, the vendor will should:

Provide built-in DoS resiliency for all components processing signaling and audio.

Provide embedded or compatible third party firewalls, IDS/IPS systems, or anti-DoS systems will be available.

Support DoS detection and mitigation capabilities in network switches Provide a solution for malformed or “fuzzed” packets Provide protection for key supporting infrastructure services, such as

TFTP, DHCP, DNS, etc. will be provided. Vendor Response Requirements Briefly describe any embedded features/functions in the proposed IPTS solution that will reduce probability of telephony services disruption due to Denial-of-Service (DoS) attacks and address each of the above listed items in your response.

Cisco Response: While the most effective methods for stopping DoS attacks are deployed in the infrastructure, Unified Communications Manager utilizes a



number of tactics to manage DoS exposure. Unified Communications Manager 7.0 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 Communications Manager 7.0 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 Communications Manager servers and only permit clients to connect on certain ports/sockets. For example, native database access is only provided within/between the Unified Communications Manager 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 Communications Manager 7.0 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 Communications Manager. 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 Communications Manager, the wide array of redundancies Unified Communications Manager affords you makes it nearly impossible for an attacker to find a silver bullet. Signaling can be handled by multiple Unified Communications Manager servers separated across a network, so if one were to be attacked the others could fill its place. If a Unified Communications Manager were to be taken down by an attack, remote SRST gateways could take over signaling for branches. Unified Communications Manager 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.

1.6.4 Theft of Services Vendor Response Requirements Briefly describe any embedded features/functions in the proposed IPTS solution that will identify the incidence of toll fraud and other types of Long Distance toll service abuse/misuse (e.g. LD voice calls on fax lines) in real-time, and alert and/or block such activity to reduce financial losses.

Cisco Response: Unified Communications Manager provides a rich set of Toll fraud counter-measures. These capabilities are integral to the Unified Communications Managers 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.

1.6.5 Restricted Calls Vendor Response Requirements Briefly describe any embedded features/functions in the proposed IPTS that will identify telephony/fax spam, harassing calls, and other types of restricted calls (e.g. bomb threats, threatening calls, calls to/from restricted numbers) in real-time, and alert and/or block such activity to reduce damages and legal exposure.

Cisco Response: Malicious Call Identification (MCID), an internetwork service, allows users to initiate a sequence of events when they receive calls with a malicious intent. The user who receives a disturbing call can invoke the MCID feature by using a softkey or feature code while connected to the call. The MCID service immediately flags the call as a malicious call with an alarm notification to the Cisco Unified Communications Manager administrator. The MCID service flags the call detail record (CDR) with the MCID notice and sends a notification to the off-net PSTN that a malicious call is in progress. The system supports the MCID service, which is an ISDN PRI service, when using PRI connections to the PSTN.



1.6.6 Confidentiality and Privacy (Packet Sniffing)

VoiceCon requires that the proposed IPTS solution provide for a high degree of confidentiality and privacy, including:

Support for standards such as IPSec, TLS, and SRTP. Encryption for all public (to the LAN) traffic must be supported. This

includes traffic exchanged between the call processor and media gateway.

Vendor Response Requirements Briefly describe any embedded features/functions in the proposed IPTS that will preserve communications confidentiality and privacy, including the standards listed above. 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 Communications Manager, 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 Communications Manager 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. 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 Communications Manager 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 Communications Manager 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 Communications Manager 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.

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

Cisco Response: Each Unified Communications Manager server supports a single physical interface for administration, control, and voice signaling (or two interfaces in a failover configuration). However, there are a number of features native to Unified Communications Manager 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 Communications Manager 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.

1.6.7 Root Access

Vendor Response Requirements Is there direct Root access to the IPTS common control, and does the proposed IPTS solution conform to the following design attributes:

Disablement of non-secure management interfaces such as telnet by default.

No installation of any default administrative or root passwords. Logging of all activity for administrative or root access.

Cisco Response: Cisco Unified Communications Manager 6.1 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 Communications Manager Release 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.

1.6.9 Miscellaneous Security Requirements Vendor Response Requirements VoiceCon requires that the proposed IPTS solution provide the following general security features:

A patch management process and system must be available. A secure alternative to TFTP (whose files can easily be sniffed) must be

provided. Support of TCP and authentication should be provided if SIP is

supported. Firmware loads for IPTS phones will be signed to insure authenticity.

Vendor Response Requirements Confirm that the proposed IPTS solution satisfies each of the listed general security features:

Patch management TFTP alternative Signed firmware loads

Cisco Response: The security of Cisco Unified IP Phones begins with signed images. This feature allows a 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 Communications Manager 3.3(3), all images are signed by default with no configurable options. Cisco Unified IP Phones also support authentication and verification of integrity for configuration files using public-key digital signatures. Communications Manager 5.0 introduced configuration file encryption using AES-128-CBC (Cipher-Block Chained). Although the TFTP standard doesn’t provide any security features, it doesn’t mean the use of TFTP is insecure. We’ve applied industry standard practices for authentication, integrity, and encryption at the application layer to circumvent the drawbacks to TFTP’s security.



1.7 Green Initiatives VoiceCon is committed to supporting the environment and is currently following a “Green” corporate strategy. The selected IPTS solution must confirm to this strategic objectives 1.7.1 Power Management 1.7.1.1 Energy Savings Vendor Response Requirements How does the currently proposed IPTS solution save on required energy costs compared to earlier TDM-based PBX models and/or IP-based versions? Specifically address energy savings (reduced power requirements) as regards any or all of the following system elements:

Common Control Switching Network Common Equipment, including cabinets/carriers, circuit boards (service

circuits, port circuits, media gateways, et al) Desktop telephone instruments Voice messaging system

Cisco Response: Cisco Green Vision: At Cisco, we’re committed to changing the way people work, live, play, and learn. And we believe this vision extends to our ability to impact the environment in a positive way. Together with our customers, partners, employees, and communities, we have a unique opportunity to help everyone use technology in an environmentally sustainable way. Before green was big business, Cisco was already focused on developing products that were energy-efficient. Our commitment to sustainability extends to all phases of our products—from design and manufacturing through support and end of life. Our green product design considerations include energy consumption, materials selection, packaging, upgradeability, and recyclability.

Because the vast majority of energy consumption is attributed to the data center and other high-end network equipment, we are committed to developing advanced, energy-efficient data center architectures at the systems level.

Cisco Unified Communications - Specifically, in the area of unified communications we are focusing on improving scalability and co-residency of applications to reduce the number of servers required for complete systems. Reducing server count through improving the scalability of individual servers, making applications co-resident, or through implementing virtualization directly contributes to reduce energy costs.

Examples:

Cisco Unity 7.0 now supports 15,000 users and 200 ports on a single MCS 7845 server. This compares to 7.500 users and 144 ports for Unity 5.0.



Cisco Unified Communications Manager 5.0 (and higher) includes Mobility Manager, so a separate server is no longer required for that application.

Cisco UC Integration for MOC allows customers to greatly reduce their server count by eliminating the need for Mediation Servers in the Microsoft OCS solution.

Cisco Unified Communications Manager 7.0 Business Edition combines call processing and unified messaging and mobility on a single-server solution for up to 500 users.

Cisco is working on reducing power usage of IP phones while simultaneously providing the best in class IP phones. The newer versions in the Cisco Unified IP Phone portfolio include devices with more features while reducing overall power consumption by 41%.

Another simple but effective measure also involves the addition of automatic shutdown capabilities in many of our high-volume products, such as telephones, set-top boxes, and modems. For example, the color display phones have a “screen saver” mode to extend screen life and save power, and they can be set to go into this mode in off-hours by the user or the administrator. In addition, we are looking at ways to power down the phones during idle hours, but this must be done very carefully since the phones must have “instant-on” capability as well.

Power over Ethernet Considerations In March of 2000, well ahead of the Power over Ethernet industry standards, Cisco released the first set of switches and phones that used PoE. In June 2003 the IEEE ratified the 802.3af specification, which defined the standard for device detection and delivery of PoE. This standard specifies various power classes as shown in the following table:

802.3af Classifications

802.3af Class

Maximum Power (W)

Class Description

3 15.40 Full-power

2 7.00 Medium-power

1 4.00

Low-power

0 15.40 Unknown

The PoE class value is used to indicate the peak power usage that will be consumed by devices. As defined in the standard, PoE does not “negotiate” power requirements. Instead, it uses resistance values and current to determine peak power consumption for a given device. However, the 802.3af classifications are inflexible since peak power usage is based on classification instead of negotiation. After initial device power up, power requirements cannot be adjusted. Consider a device that has a Class 3



rating. The power sourcing equipment (PSE) will allocate 15.4 watts (W) of power even though the device may actually only consume 7.01W of power, representing an over-budgeting of twice the amount of power actually needed. This over allocation would result in the PSE supporting significantly fewer devices than it has available power capacity for. In this example, a Cisco switch or router would be able to save 50% of the power consumed on a port when connected to a Cisco Unified IP Phone.

Cisco Discovery Protocol Power Negotiation When deploying a small number of IP phones, the over budgeting of the power requirements based on a device’s class would not be significant. But in large deployments it could be a significant factor. To more effectively manage power budgeting for Cisco Unified IP phones connected to a switch, the Cisco Discovery Protocol communicates an accurate power requirement value to the connected switch. In fact, select Cisco switches are capable of delivering power to other switches, wireless Access Points, IP cameras, PoE speakers, PoE digital clocks, and personal computers. The use of Cisco Discovery Protocol does not affect the 802.3af discovery process, but instead downwardly adjusts the amount of power required by the device and allows additional devices to be supported. As an example, the Cisco Unified IP Phone 7911G is an AF Class 2 device. AF Class 2 devices consume a maximum of 7.0W of power from the PSE. After the Cisco Unified IP Phone 7911G powers up, it exchanges Cisco Discovery Protocol with a Cisco switch and the phone reports maximum power consumption at 5.0W. This is an immediate 29 percent savings in the PSE’s power budget. When measuring the actual power consumption of the Cisco Unified IP Phone 7911G after it has registered with a call processing server (using the factory defaults), the phone consumes only 2.3W of power. This represents a 54 percent difference between the Cisco Discovery Protocol reported value and the actual usage, and a 68 percent difference from the AF classification. Real-Life Deployments and Cisco IP Phone Power Usage Precisely how much power a Cisco Unified IP Phone consumes will depend on both the phone model as well as individual user settings. Phone model/features and user preferences that can increase power consumption on a Cisco Unified IP Phone are: screen type (color versus grayscale), Gigabit Ethernet line speed, ringer/speakerphone volume, and illuminated keys (e.g. Message Waiting light, lines buttons). When looking at the power consumption of a Cisco Unified IP Phone, the amount of power used daily will be reflected predominantly in the idle power usage of the phone. Although ringing will temporarily increase power usage, over the course of the day, ringing will not statistically affect the overall power consumption of the phone. On a few of the Cisco Unified IP Phones there is a slight power usage increase when there is activity on the handset/headset, but when calculating the daily power usage for the phone, the active call and ringing increases will merely add approximately 1 percent to the idle power usage. Although each user configures and interacts with their phone differently, the overall power usage during the day remains relatively flat. The total power draw including ringing and talk time has less than a one percent deviation from idle. This daily usage calculation assumes six incoming calls and talk time of 3.5 hours. If power consumption is based on the 802.3af power allocation or the



Cisco Discovery Protocol value, the Cisco Unified IP Phone 7975G would be estimated to consume 370 watts or 288 watts respectively. However, because the idle power draw is a more accurate reflection of the overall power consumption, the actual power consumption is only 92 watts.

1.7.1.2 Reduced Cooling Costs Vendor Response Requirements How does the currently proposed IPTS solution save on required cooling costs by reducing BTU dissipation as compared to earlier TDM-based PBX models and/or IP-based versions? Please be specific as to BTU dissipation by hardware element, if possible.

Cisco Response: As noted above, reduced server count contributes to reduced BTU dissipation. In addition, new Cisco gateways based on Cisco Integrated Services Routers utilize higher density modules (For PRI, T1, FXS, FXO, etc.) in the same energy efficient chassis. The main power draw and BTU production for the system is generally attributable to the PoE Ethernet switches and the IP phones. As noted above Cisco has reduced the power consumption (and thus BTU dissipation) of the newer IP phones by 41% compared to earlier comparable models.

1.7.2 ISO Standards Vendor Response Requirements Briefly describe how your company is conforming to the ISO 14000 family of environmental management standards as it relates to the proposed IPTS solution? Topics for discussion may include manufacturing process, shipping, field installation and service technicians, materials recycling, documentation, et al.

Cisco Response: Cisco's ISO 14001 Environmental Management System (EMS) provides a continuous cycle of planning, implementing, reviewing, and improving the processes and actions that are performed to meet business and environmental goals. It influences all aspects of Cisco's operations, products, and services, including compliance with environmental requirements and regulations, in addition to driving ongoing improvements to environmental performance.

The EMS enables Cisco to conform to ISO 14001, which is an internationally accepted standard for environmental management systems. Cisco utilizes ISO 14001 to drive its environmental performance and compliance with environmental regulations and requirements, while continually improving the EMS. Conformance to the standard is driven by corporate-level processes, but is implemented on a site-specific basis, depending on local, theater-based, or site-related environmental concerns. Certification to ISO 14001 is performed by independent, third-party auditors.



Cisco successfully passed its first ISO 14001 registration audit in November 2000. In an effort to meet customer requests and support our ongoing commitment to the environment, Cisco continues to expand and improve its ISO 14001 certification. Cisco's ISO 14001 certified sites, including their certification year, is shown below.

Cisco ISO 14001 Certified Sites:



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, central office trunk circuits (analog and digital) for local and long distance services, and private network IP trunk circuits). 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 Headquarters Facility The Headquarters location is a two building campus. Each building is a four-floor facility, with station equipment uniformly distributed within and across the four floors of each building. There are five (5) wiring closets per floor and one (1) main equipment room on the first floor of each building. 2.1.2 Regional Office Facility The RO facility has two floors with station equipment uniformly distributed within and across each floor of the building. There are five (5) wiring closets per floor and one (1) main equipment room on the first floor. 2.1.3 Branch Office Facility The BO facility has one floor with uniformly distributed station equipment across building. There are two (2) wiring closets and one (1) main equipment room. 2.1.4 Satellite Office Facility The SO facility has one floor with uniformly distributed station equipment across the building. There is one (1) wiring closet/main equipment room. 2.2 Port Requirements VoiceCon requires that the proposed IPTS communications solution be designed to support the following equipped port capacity requirements in the following tables with wired capacity for 50% growth at the Headquarters campus facility. The definition of “equipped capacity” is all necessary system hardware and software components installed and working at time of initial system installation based on the stated requirements of this RFP. The definition of “wired for capacity” is the capability for system expansion (call processing, switched connections, and port capacity) with the addition of port interface cards or media gateway boards, only.

Cisco Response: Cisco complies. The proposed Cisco IP telephony solution easily supports the initial deployment of 2,000 users, and can scale to 3,000 users (50% growth) without any change or additions to the solution as proposed. Additional servers and 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.



VoiceCon will engineer its WAN trunk circuits to support compressed voice traffic (G.729A algorithm voice codecs) across its facilities. In addition to the following trunk circuit requirements any additional trunk services necessary to support the proposed IPTS solution, e.g. local survivability requirements, must be identified, configured, and included in the pricing proposal. Necessary common equipment must be included in the system configuration and pricing proposals and identified as such.

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

2.2.1 Equipped CHQ Station/Trunk Port Requirements 2.2.1.1 Building 1 Station Equipment

Analog devices (VoiceCon provided): 75 o 2500-type telephone instruments: 55 o Modems: 15 o Facsimile terminals: 5

IP terminals (See Section 4) 675 o Economy desktop instrument 25 o Administrative desktop instrument 50 o Professional desktop instrument 350 o Executive desktop instrument 225 o Attendant soft consoles 3 o Audio conferencing units 22

TDM Trunk Circuits Local Service

o GS/LS circuits 48 o T1-carrier (PRI) circuits 3

Long Distance o T1-carrier (PRI) circuits 2

2.2.1.2 Building 2 Station Equipment


IP terminals (See Section 4) 675 o Economy desktop instrument 25 o Administrative desktop instrument 50 o Professional desktop instrument 350



o Executive desktop instrument 225 o Attendant soft consoles 3 o Audio conferencing units 22




2.2.2 Equipped RO Station/Trunk Port Requirements Station Equipment



Trunk Circuits Local Service



2.2.3 Equipped BO Station/Trunk Port Requirements Station Equipment



Trunk Circuits



Local Service o GS/LS circuits 8 o T1-carrier (PRI) circuits 1


2.2.4 Equipped SO Station/Trunk Port Requirements Station Equipment






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 location.

Location T1 (local Inbound/ Outbound)

T1 Long Distance

Analog FXS

2-way GS/LS FXO

CHQ1 3 3* 96* 48 CHQ2 3 3* 96* 48 RO 2 2 28* 20 Branch 1 1 12* 8 Satellite 0 0 8* 8

*Indicates that the quantity included exceeds the required amount.



Note: The GS/LS emergency trunks have been provisioned as FXO trunks on the gateway routers, with 8 ports at each site provided via the Gordon Kapes units.

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 two basic gateway options:

1. Gateway modules for Cisco Integrated Services Routers, which include FXS, FXO, T1, E1, BRI, PRI, TIE, and CAMA.

2. 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, the following gateways have been included in the proposal:

1) Cisco Integrated Services Routers (ISRs) model 2811 with VWIC, or WIC or high density modules for T1, FXO or FXS ports.

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

Note: A complete overview and detailed descriptions of the Cisco Integrated Services Routers and VG248 Analog Phone Gateways can be found at:

Cisco Integrated Services Routers: http://www.cisco.com/en/US/products/hw/routers/index.html#~hide_v3~+hide-id-trigger-g1-branch VG248 Analog Phone Gateways: http://tools.cisco.com/search/JSP/search-results.get?strQueryText=vg248+analog+phone+gateways&Search+All+cisco.com=cisco.com&language=en&country=US&thissection=f&accessLevel=Guest



3.0 Port Interface and Traffic Handling Requirements The proposed IPTS must support a variety of peripheral ports and switched connections. In addition to IP endpoints it is required to support traditional circuit switched analog stations and analog & digital trunk circuit interfaces. The common equipment (port interface carriers, media gateways) must be supported in a distributed topology using VoiceCon’s LAN/WAN for transmission and switching of communications and control signaling. Common equipment must be located at CHQ, RO, and SO facilities. Any and all port interface cabinets/carriers designed to support traditional analog and digital interface ports should include an integrated TDM bus backplane traffic engineered to support non-blocking switch network access for all peripheral endpoint connections. Transmission and connections between all TDM buses must also be traffic engineered to operate in non-blocking mode. A center stage switch network, if equipped, must also be traffic engineered for non-blocking access. Media gateway equipment should be designed and configured to support a 4:1 ratio between IP peripheral endpoints (line station and trunk circuit) and media gateway channels used to connect to non-IP ports. For common equipment configuration design purposes assume the following voice communications traffic volumes: User line station (analog and IP) 12 CCS at busy hour Attendant console position 36 CCS at busy hour ACD/Supervisor call center agents 36 CCS at busy hour Voice mail port traffic 36 CCS at busy hour All trunk circuit traffic 36 CCS at busy hour The assumed system voice communications traffic mix should be 40% station to station calls, 30% incoming trunk calls, 30% outgoing trunk calls. Vendor Response Requirement: The proposed system IPTS be designed and engineered 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 10/100/1000 Mb switched Ethernet LAN infrastructure. All end points connect to a 10/100/1000 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 Communications Manager server in a non-blocking configuration, and up to 30,000 Unified IP Phones in a Unified Communications Manager cluster.

3.1 Common Equipment 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 and PSTN trunk circuits. Switched connections involving non-IP ports may be handled using a circuit switched network, media gateways/Ethernet switches, or a combination of both. Vendor Response Requirement: Briefly identify by make/model the proposed common equipment and describe each type of equipment housing used to support port circuit interface card, media gateway boards, and other required equipment. Specifically discuss in the response housing size (H x L x W), weight (standard common assembly), fan cooling units, power supply requirements, number of usable port card slots per carrier/chassis, and all embedded hardware components.

Cisco Response: Cisco utilizes Integrated Services Routers (ISRs) to provide TDM gateway services. The physical characteristics of the ISRs proposed in this RFP (Cisco 2811 ISRs) are listed in the tables shown in Sections 1.3.1.1.1 and 1.3.5.3 above. Further details about the hardware components, usable slots, etc., can be found at: http://www.cisco.com/en/US/products/hw/routers/index.html#~hide_v3~+hide-id-trigger-g1-branch

3.1.1 Universal Card Slots VoiceCon prefers that the proposed common equipment be based on a universal card slot design for all TDM port interface circuit cards. Vendor Response Requirement: Confirm that your proposed system satisfies this requirement.

Cisco Response: Cisco utilizes Integrated Services Routers (ISRs) to provide TDM gateway services. These ISRs have universal module slots for Cisco network modules to support a wide variety of port types (FXS, FXO, TIE, T1/E1 PRI, etc.). http://www.cisco.com/en/US/products/hw/routers/index.html#~hide_v3~+hide-id-trigger-g1-branch

3.1.2 Common Equipment Redundancy VoiceCon requires an IPTS that satisfies a very high degree of reliability and services availability. To achieve this goal IPTS common equipment should



include a significant number of redundant design elements to minimize the affects of single points of failure.

Cisco Response: A full description of the redundant common control elements is included in Section 1.3.1.1 above.

Vendor Response Requirement: Confirm if the proposed common equipment includes any or all of the following redundant common equipment elements, and indicate type of redundancy provided for each element. Common Equipment Element Duplicated, Load Sharing, or N/A Service Circuits Cards TDM Port Interface Circuit Cards I/O Interfaces Media Gateway Boards TDM bus backplane Inter-TDM bus connections Center-stage Circuit Switch Power Supplies

3.2.1 IP Station Discovery Vendor Response Requirement: 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 specifications for in-line power of IP telephone equipment. If 802.3af is not supported, identify the PoE implementation being proposed.

Cisco Response: The Cisco Unified IP phones specified in this RFP support IEEE 802.3af in-line power.

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 Handling Multi-Party Conference Calls The proposed system must be able to support six party add-on conference calls among IPTS and off-network stations with a minimum of three (3) off-network stations per call when required. The IPTS solution must also support a minimum of 20 simultaneous multi-party conference calls (up to six parties per call and stations located anywhere across the VoiceCon network and/or off-network per call) 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 server, is required.

Cisco Response: Cisco Unified Communications Manager 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 Communications Manager cluster at Headquarters as well as the DSP modules in the ISR’s in the Regional Office and branch offices.

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

Software conference devices support a variable number of audio streams. You can create and configure a software conference device within a Unified Communications Manager 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 Communications Manager 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 Handing VoIP Overflow Traffic If available WAN circuits connecting VoiceCon facilities are busy, call admission control levels are reached, or QoS levels are not satisfied 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 Communications Manager 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 Station Endpoints (desktop telephone instrument, PC client soft phones 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 describe how call control signaling is routed (identifying any and all circuit card and transmission bus requirements).

Cisco Response: Cisco Unified IP Phones are logically supported by Cisco



Unified Communications Manager 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 Telephone Instrument Vendor Response Requirement: Provide a brief description how analog telephones are logically and physically supported by the common control call telephony server, identifying all hardware elements necessary for control signaling transmission. Specify the number of circuit terminations/bearer channels per proposed circuit board, module, or media gateway. Provide information for each of the following analog devices:

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)

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 hardware elements necessary for control signaling transmission. Specify the number of circuit terminations/bearer channels per proposed circuit board, module, or media gateway.

Cisco Response: Analog Fax support is provided by the gateways listed in 3.5.2 above.

3.5.4. Modem Vendor Response Requirement: Provide a brief description how modems are logically and physically supported by the common control call telephony server, identifying all hardware elements necessary for control signaling transmission. Specify the number of circuit terminations/bearer channels per proposed circuit board, module, or media gateway. Provide information for each of the following analog devices:



Cisco Response: Modems are supported by the gateways listed in 3.5.2 above.

3.5.5 Power Failure Transfer Station (PFTS) Vendor Response Requirement: Provide a brief description how Power Failure Transfer Stations using analog telephone instruments are logically and physically supported by the common control call telephony server, identifying all hardware elements necessary for control signaling transmission. Specify the number of circuit terminations/ bearer channels per proposed circuit board, module, or 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 Regional and satellite offices.

3.5.6 GS/LS CO Trunk Vendor Response Requirement: Provide a brief description how GR/LS trunk circuits are logically and physically supported by the common control call telephony server, identifying all hardware elements necessary for control signaling transmission. Specify the number of circuit terminations/bearer channels per proposed circuit board, module, or 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, 2 or 4 ports) Cisco High Density Expansion Modules for Cisco Routers (3 or 8 ports)

3.5.7 ISDN PRI/T1-Carrier Interface Trunk Vendor Response Requirement: Provide a brief description how ISDN PRI services carried over T1-carrier trunk interface circuits are logically and physically supported by the common control call telephony server, identifying all hardware elements necessary for control signaling transmission. Specify the number of T1-circuits per proposed circuit board, module, or 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) 3.5.8 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 hardware elements necessary for control signaling transmission. Specify the number of circuit terminations per circuit board, module or 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., are covered in detail at the following Cisco web site: http://www.cisco.com/en/US/prod/routers/networking_solutions_products_genericcontent0900aecd806cab99.html



4.0.0. Voice Terminal Instruments The proposed communications system must be able to support a mix of desktop analog and IP telephone instruments (including audio conferencing units), PC client soft phones, attendant soft consolses, and mobile communications devices. VoiceCon will provide its own analog telephone instruments, fax terminals, and modems. 4.1 Regulation Requirements All single- and multi-line 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 regulation 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

Vendor Response Requirement: In a separate PPT file attachment provide a slide illustration (graphic or photograph) of the four proposed desktop IP telephone instruments with model names identified. Include in the illustration any add-on modules or options required to satisfy individual model requirements.

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 7962G plus 7915

Key Expansion Module Professional IP Phone: Cisco Unified IP Phone 7965G Executive IP Phone: Cisco Unified IP Phone 7975G Audio Conference Station: Cisco 7937G IP Conference Station PC Client Softphone Cisco Unified Personal Communicator



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

12-key dial pad Hold button G.711/G.729 voice codecs Auto Self Discovery/DHCP QoS Support (802.1p/Q, DiffServ) Echo Canceller Support IEEE 802.af POE specifications

Vendor Response Requirement: Clearly state your proposed economy model and confirm that it fully satisfies each of the stated requirements. Also provide a brief description of the proposed instrument (including supported communications protocols) and identify any worthwhile attributes/capabilities that exceed the stated requirements, e.g. speakerphone, display, programmable line/feature key, et al.

Cisco Unified IP Phone 7906G

Cisco Response: The Cisco Unified IP Phone 7906G has been quoted to meet this requirement. It meets all of the requirements listed above. 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 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 Communications Manager 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 Communications Manager 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 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 Can optionally support an add-on key module (12 line/feature keys,



minimum) with soft label/status indicators G711, G729 and wideband (G.722 or equivalent) 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 or slide High resolution, backlit, monochrome grayscale graphical display screen

with four (4) associated context sensitive soft feature keys Cursor/navigator interface for display screen interactionl LDAP access Stored Call Data (Last 25 numbers dialed; Last 25 incoming call

numbers; Last 25 missed call numbers) Integrated Ethernet switch with two (2) RJ-45 connector interface ports

for 10/100 Mbps LAN and desktop PC connectivity Bluetooth headset interface (DECT interface also acceptable) Support of IEEE 802.af POE specifications

There is also a requirement for the Administrative model to support SIP specifications as either its standard or optional communications signaling protocol. Vendor Response Requirement: Clearly state your proposed Administrative model and confirm that it fully satisfies each of the stated requirements. Also provide a brief description of the proposed instrument (including supported communications protocols) and identify any worthwhile attributes/capabilities that exceed the stated requirements. Specifically identify any requirement not fully satisfied, e.g., soft feature key substituted for fixed feature key requirement. Complete the following table,




Cisco Response: The Cisco Unified IP Phone 7962G has been quoted to meet this requirement in combination with the Cisco IP Phone Expansion Module 7914.

The Cisco IP Phone 7962G, an additional offering in the IP Phone portfolio, is an advanced, fully-featured manager IP designed to provide users with hi-fidelity, life-like voice communications. A high-quality hands-free speakerphone and handset designed specifically for wideband/G.722 audio are standard, as is support for wideband headsets. The Cisco Unified IP Phone 7962G is intended to meet the needs of managers and administrative assistants. It provides six programmable backlit line/feature buttons and four interactive soft keys that guide a user through all call features and functions. The new functionality and features are integrated within an industry-proven, award-winning industrial design.

This state-of-the-art IP phone includes a crisp, 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 innovative and productivity-enhancing, higher value, and more visibly rich Extensible Markup Language (XML) applications to the display. Double-byte languages are also supported on the Cisco IP Phone 7962G. Dynamic backlit 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 7962G does not have a built-in Bluetooth headset adapter or a USB port. Features

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

Directories --The Cisco IP Phone 7962G 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.

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), CallManager software, and backup CallManager software.

Services -- The Cisco IP Phone 7962G 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 IP Phone 7962G features high-quality wideband-capable 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 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.

The dial pad is also ADA-compliant.

The foot-stand of the Cisco IP Phone 7962G 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 IP Phone 7962G supports up to two Cisco IP Phone Expansion Module 7914 units.

Other Cisco Unified IP Phone 7962G features include:

24+ user-adjustable ring tones

G.711a, G.711µ, G.729a, G.729ab, G.722 and iLBC audio compression codecs are supported

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, Audio Preferences and Contrast

Network Configuration

Device Configuration

Security Configuration



Cisco Unified IP Phone Expansion Module 7915 (shown attached to Cisco Unified IP Phone)

Enhanced Call Coverage Capabilities

Call coverage is a critical capability for executive administrators and others who monitor and manage the various status of calls. For high call volume positions, the Cisco Unified IP Phone Expansion Module 7915 and the Cisco Unified IP Phone Expansion Module 7916 deliver an adjunct solution to instantly determine the status of a number of lines beyond the number supported natively in the wideband audio Cisco Unified IP Phones 7962G, 7965G and Phone 7975G. The Cisco Unified IP Phone Expansion Modules extend the call coverage capabilities with additional buttons and an LCD display.

Key Features of Cisco Unified IP Phone Expansion Module 7915 and 7916

The large, high-resolution LCD display allows for quick and easy identification of associated buttons. The Cisco Unified IP Phone Expansion Module 7915 has a grayscale display; the Cisco Unified IP Phone Expansion Module 7916 has a color display.

The Cisco Unified IP Phone Expansion Module 7915 and 7916 feature 12 physical keys with access to 12 additional keys via the page keys. This adds a total of 24 buttons to the existing six buttons of the Cisco Unified IP Phones 7962G and 7965G and the existing eight buttons of the Cisco Unified IP Phone 7975G.

Up to two Cisco Unified IP Phone Expansion Modules 7915 or 7916 can be used with Cisco Unified IP Phones 7962G, 7965G, or 7975G.

Users can adjust the contrast or brightness of the individual LCDs for the Cisco Unified IP Phones 7962G, 7965G, and 7975G and Cisco Unified IP Phone Expansion Module 7915 and 7916 according to their preference.

The buttons on each Cisco Unified IP Phone Expansion Module can be programmed as a directory number (DN), line key, or speed-dial key, much like the Cisco Unified IP Phones 7962G, 7965G, and 7975G. When used as a DN key, buttons are illuminated, allowing easy identification of call state.

Attributes Yes/No 12-key dial pad Yes



16 programmable line/feature keys with soft label/status indicators

Yes

Add-on key module (12 line/feature, soft label/status indicators, minimum) option

Yes

G711, G729 and wideband voice codecs Yes Auto Self Discovery/DHCP Yes Echo Canceller Yes QoS Support (802.1p/Q, DiffServ) Yes

Hold key Yes (Soft Key)

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

Yes

Four (4) associated context sensitive soft feature keys Yes Cursor/navigator interface for display control Yes LDAP access Yes Stored Call Data (Last 50 numbers dialed/Last 50 incoming call numbers/Last 50 missed calls)

Yes

Integrated 10/100 Mbps Ethernet switch (2 RJ-45 connector ports)

Yes

Bluetooth/DECT headset interface (indicate which type) No IEEE 802.af POE support Yes 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 (G.722 or equivalent) voice codecs Auto Self Discovery/DHCP Echo Canceller QoS Support (802.1p/Q, DiffServ) Embedded Web services support, e.g., XML 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 graphical display screen

with four (4) associated context sensitive soft feature labels ((key,



cursor, or navigator control) LDAP access Stored Call Data (Last 50 numbers dialed/Last 50 incoming call

numbers/Last 50 missed calls) Integrated (embedded or add-on) Gigabit Ethernet switch with two (2)

RJ-45 connector interface ports for 10/100/1000 Mbps LAN and desktop PC connectivity.

Bluetooth headset interface (DECT interface also acceptable) Support of IEEE 802.af POE specifications

There is also a requirement for the Professional model to support SIP specifications as either its standard or optional communications signaling protocol. Vendor Response Requirement: Clearly state your proposed Professional model and confirm that it fully satisfies each of the stated requirements. Also provide a brief description of the proposed instrument (including supported communications protocols) and identify any worthwhile attributes/capabilities that exceed the stated requirements. Specifically identify any requirement not fully satisfied, e.g., soft feature key substituted for fixed feature key requirement. Complete following table:

Cisco IP Phone 7965G

Cisco Response: The Cisco IP Phone 7965G has been quoted to meet this requirement. The Cisco Unified IP Phone 7965G is an advanced, color, fully-featured business IP phone designed to provide users with rich, hi-fidelity, life-like voice communications. A high-quality hands-free speakerphone and handset designed specifically for wideband/G.722 audio are standard, as is support for wideband headsets. The Cisco Unified IP Phone 7965G is intended to not only meet the needs of professional workers, managers and administrative assistants, but also those of power users/developers working



with bandwidth-intensive applications on co-located PCs. Addtionally, users that require/prefer vibrant color displays for useability reasons or for productivity-enhancing applications will benefit from this phone. It provides six programmable backlit line/feature buttons and four interactive soft keys that guide a user through all call features and functions. The new functionality and features are integrated within an industry-proven, award-winning industrial design.

This state-of-the-art IP phone includes a backlit TFT color display (320 x 240) for easy access to communication information, timesaving applications, and feature usage. It also enables customers and developers to deliver innovative and productivity-enhancing, higher value, and more visibly rich Extensible Markup Language (XML) applications to the display. Double-byte languages are also supported on the Cisco IP Phone 7965G. Dynamic backlit tri-color buttons provide straightforward call state identification. Both local powering options and IEEE 802.3af PoE (Class 3) are supported.

Features


Directories -- The Cisco IP Phone 7965G 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), Communications Manager software, and backup Communications Manager software.


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.


The internal Cisco 2-port Ethernet switch allows for direct connections to a 10/100/1000 BASE-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 IP Phones, providing improved security and reliability of voice and data traffic.





The dial pad is also ADA-compliant.

The foot-stand of the Cisco IP Phone 7965G 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.






Enhanced navigation with addition of 4-way navigation button plus ‘Select’ key




User Preferences: Ring Tones, Background Images, Audio Preferences and Brightness




Attributes Yes/No 12-key dial pad Yes 6 programmable line/feature keys with soft label/status indicators

Yes

G711, G729 and wideband voice codecs Yes

Auto Self Discovery/DHCP Yes Echo Canceller Yes QoS Support (802.1p/Q, DiffServ) Yes Embedded Web services support (indicate specification) Yes

Hold key Yes (soft key)

Last Number Redial key Yes Release key Yes Message Waiting/Call Ringing indicator(s) Yes Full Duplex Speakerphone Yes Speaker/Mute key Yes Volume Control keys/slide Yes



High resolution, backlit, monochrome grayscale pixel display

Yes (Color)

Four (4) associated context sensitive soft feature keys Yes Cursor/navigator interface for display control Yes LDAP access Yes Stored Call Data (Last 50 numbers dialed/Last 50 incoming call numbers)

Yes

Integrated 10/100/1000 Mbps Ethernet switch (2 RJ-45 connector ports)

Yes

Bluetooth or DECT headset interface (indicate type) No IEEE 802.af POE support Yes 4.2.4 Executive Desktop IP Telephone Instrument The Professional model will be used by VoiceCon ’s executive management team and select managers. It should have, at minimum the following design attributes and features/functions:

12 key dial pad Ten (10) programmable line/feature keys with soft label/ status

indicators G711, G729 and wideband (G.722 or equivalent) voice codecs Auto Self Discovery/DHCP Echo Canceller QoS Support (802.1p/Q, DiffServ) Embedded Web services support, e.g., XML 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 labels (key, cursor, or navigator control) with a touch screen interface

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

numbers/Last 50 missed calls) Integrated (embedded or add-on) Gigabit Ethernet switch with two (2)

RJ-45 connector interface ports for 10/100/1000 Mbps LAN and desktop PC connectivity.

Bluetooth headset interface (DECT interface also acceptable) Support of IEEE 802.af POE specifications

Vendor Response Requirement: Clearly state your proposed Executive model and confirm that it fully satisfies



each of the stated requirements. Also provide a brief description of the proposed instrument (including supported communications protocols) and identify any worthwhile attributes/capabilities that exceed the stated requirements. Specifically identify any requirement not fully satisfied, e.g., soft feature key substituted for fixed feature key requirement. Complete following table:


Cisco Response: The Cisco IP Phone 7975G has been quoted to meet this requirement. The Cisco IP Phone 7975G is an advanced, color, fully-featured business IP designed to provide users with rich, hi-fidelity, life-like voice communications. A high-quality hands-free speakerphone and handset designed specifically for wideband/G.722 audio are standard, as is support for wideband headsets. The Cisco Unified IP Phone 7975G is intended to not only meet the needs of executives, managers and administrative assistants, but also those of power users/developers working with bandwidth-intensive applications on co-located PCs. Addtionally, users that require/prefer vibrant color touchscreen displays for useability reasons or for productivity-enhancing applications will benefit from this top-of-the-line model. It provides eight programmable backlit line/feature buttons and five interactive soft keys that guide a user through all call features and functions. The new functionality and features are integrated within an industry-proven, award-winning industrial design.

This state-of-the-art IP phone includes a large backlit TFT touchscreen color display (320 x 240) for easy access to communication information, timesaving applications, and feature usage. It also enables customers and developers to deliver innovative and productivity-enhancing, higher value, and more visibly rich Extensible Markup Language (XML) applications to the display. Double-byte languages are also supported on the Cisco IP Phone 7975G. Dynamic backlit tri-color buttons provide straightforward call state identification. Both local powering options and IEEE 802.3af PoE (Class 3) are supported.

Features


Directories -- The Cisco IP Phone 7975G 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), Communications Manager software, and backup Communications Manager software.


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.


The internal Cisco 2-port Ethernet switch allows for direct connections to a 10/100/1000 BASE-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 IP Phones, providing improved security and reliability of voice and data traffic.



The dial pad is also ADA-compliant. The foot-stand of the Cisco IP Phone 7975G 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.






Enhanced navigation with addition of 4-way navigation button plus ‘Select’ key






User Preferences: Ring Tones, Background Images, Audio Preferences and Brightness




Attributes Yes/No 12-key dial pad Yes 10 programmable line/feature keys with soft label/status indicators

Yes

G711, G729 and wideband voice codecs Yes

Auto Self Discovery/DHCP Yes Echo Canceller Yes QoS Support (802.1p/Q, DiffServ) Yes Embedded Web services support (indicate specifications) Yes

Hold key Yes (Soft Key)

Last Number Redial key Yes Release key Yes Message Waiting/Call Ringing indicator(s) Yes Full Duplex Speakerphone Yes Speaker/Mute key Yes Volume Control keys/slide Yes High resolution, backlit, color pixel display with touch screen interface

Yes

Four (4) associated context sensitive soft feature keys Yes Cursor/navigator interface for display control Yes LDAP access Yes Stored Call Data (Last 50 numbers dialed/Last 50 incoming call numbers)

Yes

Integrated 10/100/1000 Mbps Ethernet switch (2 RJ-45 connector ports)

Yes

Bluetooth or USB interface No IEEE 802.af POE support Yes 4.2.5 Desktop IP Telephone Instrument Web Services 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 protocols (XML, HTML, WAP, et al.) supported Type of station user interaction with display screen Screen saver availability Standard embedded applications, such as visual mailbox, personal



directory/calendar, web page display, visual and/or audio alerts

Cisco Response: Cisco Unified IP phones support web-based applications using XML services and Java midlets 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, 7971G, and 7975G 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

Cisco Unified Communications Widgets: Cisco® Unified Communications Widgets is a suite of applications that deliver productive and personalized user experience with Cisco Unified Communications applications and Cisco Unified IP Phones. With these free-to-download and easy-to-add Cisco Unified Communications Widgets, you can streamline business communications and instantly access rich Cisco Unified Communications to have a tailored and familiar collaboration experience in every workspace. Phone Designer is a free-to-download Cisco Unified Communications Widget for Cisco Unified IP Phones that brings a new level of personalization to business communications in the office. With a few mouse clicks, you can quickly customize your Cisco Unified IP Phone display with wallpaper of your choice and create or change your ring tones to a preferred melody.

Cisco Unified Communications Widgets include the following three applications described in more detail below:

Phone Designer enables you to quickly customize Cisco Unified IP Phone displays with wallpapers of your choice and to create or change ring tones.

Click to Call for PCs lets you connect and collaborate with everyone instantly by allowing you to place Cisco Unified Communications Manager calls directly from your desktop productivity applications and web browsers.

Visual Voicemail enables you to view, listen, and respond to Cisco Unity and Cisco Unity Connection voicemail messages right from the Cisco Unified IP Phone display, without having to dial into your corporate voicemail box.



Phone Designer

Phone Designer: Phone Designer is a PC application that provides the following customization capabilities on Cisco Unified IP Phones through Cisco Unified Communications Manager:

Customize your vibrant Cisco Unified IP Phone color display by choosing from a variety of colorful wallpapers included with the Phone Designer application.

Import your favorite pictures into the Phone Designer application and edit them on your PC before importing them onto your Cisco Unified IP Phone.

Preview the image on your Cisco Unified IP Phone before sending the image to it.

Choose from a library of ring tones supplied with the Phone Designer application and download a unique ring tone to your Cisco Unified IP Phone.

Import an MP3 file from your music collection and import a snippet of your favorite melody as your unique Cisco Unified IP Phone ring tone.

Listen to your personalized ring tone through the Cisco Unified IP Phone speaker



Click to Call for PC’s

Click to Call for PC’s: The Click to Call application is a free-to-download Cisco Unified Communications Widget for PCs that streamlines business communications in every workspace. You can call co-workers, business partners, or customers through Cisco Unified Communications Manager without having to punch phone numbers on a Cisco Unified IP Phone. Simply highlight and click on phone numbers displayed on your desktop productivity applications or your web browser and instantly place a call to connect and collaborate with everyone.

The Click to Call application provides the following capabilities with desktop applications, Cisco Unified Communications Manager WebDialer service, and the Cisco Unified IP Phone registered to the user: Click to call from Microsoft Office Word and Excel, Microsoft Outlook,

Microsoft SharePoint, Microsoft Internet Explorer, and Mozilla Firefox applications to place a Cisco Unified Communications Manager call through the user’s Cisco Unified IP Phone

Click to call from the Personal Menu, a commonly used rapid contact information lookup capability available to Microsoft Outlook and Microsoft SharePoint users, to call one of the phone numbers associated with the contact

Automatic modification of phone numbers to place international calls, dial a co-worker’s extension, or place a call outside the enterprise to contact business partners and customers

View history of the last 10 calls made from the Click to Call application from the System Tray Menu and click to call previously dialed numbers



Visual Voicemail

Visual Voicemail: The Visual Voicemail application is a free-to-download Cisco Unified Communications Widget that delivers rich messaging experience on Cisco Unified IP Phones. You can view, listen and respond to Cisco Unity® and Cisco Unity® Connection messages right from the Cisco Unified IP Phone display without having to dial into your corporate voicemail box.

The Visual Voicemail application provides the following voice messaging capabilities for Cisco Unified IP Phones, Cisco Unified Communications Manager, Cisco Unity messaging, or Cisco Unity Connection:

Securely access a visual display of voice messages on your Cisco Unified IP Phone display screen, listed with sender name, date and message duration.

Display urgent, unheard and heard icons for each voice message on the voice message list.

Set a default voice message list to show urgent messages first, or sort it by date and time the messages were received.

Play back a voice message on your Cisco Unified IP Phone and see a real-time progress bar indicator.

Set a default voice message playback speed from the Cisco Unified IP Phone display.

Pause, rewind and forward a voice message. Reply to a sender by either calling back or sending a voicemail using

softkeys on your Cisco Unified IP Phone display. Record and send voice messages from your Cisco Unified IP Phone

display with urgent or normal priority markings.



Additional XML Applications: The following additional examples illustrate how XML applications are used in Cisco Unified Communications applications (each is illustrated with a screen shot from a Cisco Unified IP Phone):

MeetingPlace and MeetingPlace Express Phone View

Interface to Setup & Manage Meetings Initiate meetings Single button join See who is in the meeting See who is talking View list of conferences Single button join the conference Start a Reservationless meeting

Extension Mobility

Users can log into phone and it assumes their personal profile. Designed for mobile workers Also called “hot-desking” or “hoteling”



Included in Unified Communications at no additional charge Communications Manager Assistant

Cisco Unified Communications Manager Assistant improves productivity by extending such commonly used features as call filtering, immediate call diversion, and line state monitoring to administrative assistants and managers.

Intuitive manager call handling View of manager status and calls Display of call state, caller ID, and call timers Ability for managers to monitor how calls are being handled Support for up to 7,000 users, starting with Cisco Unified

Communications Manager 6.0

4.2.6 Additional Optional Attributes/Capabilities 4.2.6.1 Hardware Vendor Response Requirement: Provide a brief description of all hardware-based options not included as part of this RFP’s requirements that are currently available with any of the four proposed models. Options may include additional key modules, display modules, cordless handset, wireless LAN module, USB interfaces, et al. Indicate the specific models that support the listed hardware options.

Cisco Response: The following add-ons are available from Cisco. Cisco Unified IP Expansion Module 7914/7915/7916 - The only add-on module offered at present is the Cisco Unified IP Phone Expansion Module described above. The Cisco Unified IP Phone Expansion Module 7915 was included for the Administrative phone requirement.



Bluetooth/DECT Interface - Cisco offers a “Headset Hookswitch Control” feature to drive enhanced mobility at the desktop. This feature is available via phone firmware load 8.3(3), and requires partner vendors’ hardware. It allows a user on a wireless headset to receive ring indication remotely, plus go on-hook/off-hook and control volume/mute, all without the burden of a mechanical lifter. This is a more elegant solution for those within an office that often have reason to stray beyond the limited reach of a corded handset. It is available for the Cisco Unified IP phones 7942G, 7945G, 7962G, 7965G, and 7975G. Protocols supported include Bluetooth and DECT (DECT provides better distance and density than Bluetooth) giving partners and customers a choice of protocol and headset options. Note: This is not native wireless/Bluetooth, but requires a headset, a base station, and a data cable that runs between the base station and the Cisco Unified IP phone models 79x2 or 79x5. Since headsets were not a requirement for this RFP, no bluetooth headsets were included in this proposal. Wideband Handset - The Cisco Unified IP Phone 7962G and 7942G models, plus the Cisco Unified IP Phone 7975G, 7965G, and 7945G models support wideband across the speakerphone, handset, and headset (headset must be purchased separately). These new Cisco Unified IP Phones enhance the end-user experience with high-fidelity wideband audio and other features. In order to extend this benefit to previous Unified IP Phone models, Cisco offers wideband support via a separately orderable wideband handset (CP-WB-HANDSET=) and/or via a third-party wideband headset. This applies to the following Cisco Unified IP Phones: 7971G-GE, 7970G, 7961G-GE, 7961G, 7941G-GE, 7941G, 7911G, and 7906G. Cisco Unified Video Advantage - Cisco Unified Video Advantage (formerly Cisco VT Advantage) adds video to your communications experience by providing video telephony functionality to Cisco Unified IP phones (7900 series, and the Cisco IP Communicator softphone application). With Cisco Unified Video Advantage, video telephony is now just a phone call.

This solution comprises Cisco Unified Video Advantage software and Cisco VT Camera II, a video telephony USB camera. With Cisco Unified Video Advantage you can use the familiar phone interface to make and receive video calls on your Cisco Unified IP phone with the video component displayed on your PC. Enterprise organizations can take advantage of their existing IP networks to extend video to everyone in their organization.

When registered to Cisco Unified Communications Manager, the Cisco Unified Video Advantage-enabled phone has the features and functionality of a full-featured IP videophone. System administrators can provision a Cisco Unified IP Phone with Cisco Unified Video Advantage just as they would any other Cisco Unified IP Phone, greatly simplifying deployment and management. Enterprise customers now have a cost-effective, scalable, and visually interactive IP communications solution.

4.2.6.2 Software



Vendor Response Requirement: Provide a brief description of all software-based options not included as part of this RFP’s requirements that are currently available with any of the four proposed models. Indicate the specific models that support the listed software options.

Cisco Response: The proposed Cisco Unified IP Phones support both Cisco Skinny Client Control Protocol (SCCP) and SIP. They can be ordered initially with either protocol, and can be converted at any time with a firmware download.

4.2.7 Encryption

VoiceCon requires that its desktop IP telephone instruments support voice media and control signaling encryption using Advanced Encryption Standard (AES) or a comparable security standard Vendor Response Requirement Confirm that this requirement is satisfied for each of the four proposed models for both bearer and control transmission signals. Specify the type of encryption method deployed.

Cisco Response: Cisco Unified IP Phones (including all models quoted in this proposal with the exception of the Unified IP Conferencing Station 7937), Unified Communications Manager, 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 Communications Manager 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. Cisco encryption capabilities are described in more detail in Section 1.6.6 above.

4.3 Teleworker Options VoiceCon may require a desktop teleworker capability at some future date, be it a desktop IP telephone instrument or PC client soft phone or a combination of both.



4.3.1 Desktop IP Telephone Instrument for Teleworking A teleworker desktop IP telephone instrument should be comparable in function, capabilities, and attributes to the Professional model as described in Section 4.2.3, excluding Gigabit Ethernet. The telephone instrument should be able to connect to the host communications system via VPN, SRTP or other secure connection mechanism. Vendor Response Requirement Confirm that your proposed communications system can support a teleworker desktop IP telephone instrument that is comparable to the Professional model as described above. Identify the model(s) and provide a brief description of the proposed model. In the response specifically address each of the following:

If a standard Internet connection is acceptable for connecting to the host IPTS;

If the connection requires an external gateway of any type; Basic operational procedures required to log-in and log-out to the host

communications system; How secure communications to the host IPTS is supported; Local power requirements; If E911 calls to the teleworker’s local PSAP are supported.

Cisco Response: Cisco provides virtual private network (VPN) and voice-over-IP (VoIP) solutions that enable employees to recreate their office resources at their homes or other Internet-connected locations using a laptop or other network-accessible device, as shown in Figure 1. Using VPN services, employees can securely access data applications remotely, including e-mail, instant messaging, client/server applications, file servers, databases, and intranet services. The VPN connection can extend voicemail and employee office phone extensions directly to the employee's PC or to an IP phone.



Remote working solutions from Cisco include various combinations of data access, voice access, and high-speed transmission suitable for video conferencing and other bandwidth-intensive applications. Options include the Cisco Anywhere Office and the Cisco Enterprise-Class Teleworker solution.

The Cisco Anywhere Office

The Cisco Anywhere Office lets employees turn their company or personal laptops, public Internet terminals, or Internet-accessible wireless devices into fully functional offices from a remote location with an Internet connection. Built on Cisco remote-access VPN and Cisco Unified Communications technologies, the Anywhere Office enables access to company applications and network resources, including employee phone extensions and voicemail. The building blocks of the Cisco Anywhere Office are:

• Remote-access VPN technology provides the connection to the company network. Both types of remote access-Secure Sockets Layer (SSL) and IP Security (IPsec) are available from Cisco, integrated into a single platform, such as the Cisco ASA 5500 Series SSL/IPsec VPN Edition, or on Cisco routers. Cisco can help determine which VPN type to use for specific scenarios. Having both IPsec and SSL technology options allows businesses to customize their remote-access VPN solutions without any additional hardware or added management software.

• Security for remote workers is vital. Cisco remote-access VPN solutions allow for customization of employee access based on user device, location, endpoint security posture, and other factors. Additional security capabilities include full firewall, antivirus, anti-spyware, intrusion prevention, application control, and endpoint protection.

• Voice services extend the employee's office phone services to the remote location using Cisco IP Communicator, a Microsoft Windows-based application that allows calls to be made and received from a PC. Cisco



IP Communicator provides the same features and manageability as a Cisco Unified IP Phone handset.

The Cisco Enterprise-Class Teleworker Solution

The Cisco Enterprise-Class Teleworker solution fully replicates the office environment at a fixed, remote location. This solution provides toll-quality voice and video services and secure, robust, and centrally managed always-on high-speed VPN connections. The only hardware you need to add to the Cisco Anywhere Office is an integrated Cisco router behind the cable or DSL modem in the teleworker's home. The building blocks of the Cisco Enterprise-Class Teleworker solution are:

• Site-to-site VPN technology emulates the office environment through a Cisco router such as the Cisco 800 Series Integrated Services Router at the employee's location. Wireless LAN (WLAN) services can also be enabled on the Cisco 800 Series Software at the employee location.

• Integrated router security features from Cisco, including Cisco IOS® Firewall and Cisco IOS IPS,significantly reduce the risk of security breaches, the abuse of network privileges, and the spread of malware. Identity-based networking services provide strong authentication of users and devices to prevent unauthorized use.

• VoIP services at the employee's remote location are possible with Cisco Unified IP Phones and QoS capabilities available with Cisco IOS on the Cisco 800 Series. The same Unified Communications user experience and services available at the office-such as call routing, forwarding, conferencing, one-touch dialing, and voicemail- can be available to the remote teleworker.

4.3.2 PC Client Soft Phone for Teleworking A teleworker PC client soft phone should have comparable telephony services capabilities to the Professional model in Section 4.2.3, including the capability to function and operate as a SIP client with Microsoft Outlook compatibility. Mandatory requirements include: multiple contact directories; LDAP/Active Directory access, detailed call logs (minimum 100 last incoming, outgoing, and missed calls); click to call; virtual fixed feature keys, speed dial keys, and line keys. Vendor Response Requirement Confirm that your proposed communications system can support a teleworker PC client soft phone comparable in function to the Professional model as described above. Identify and provide a brief product description that includes an attached illustration/photograph (PPT format, only) that accurately depicts an active call screen display.

Cisco Response: Cisco has included Cisco Unified Personal Communicator 1.2 to meet this requirement. Cisco Unified Personal Communicator provides



powerful communications features integrated with your desktop or laptop 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 application and save time by not having to dial telephone numbers.

• Make calls using the integrated softphone or use Cisco Unified Personal Communicator to control your Cisco Unified IP phone on Cisco Unified Communications Manager.

• Use the Cisco Unified Personal Communicator toolbar to click to dial from within your Microsoft Outlook contacts list or e-mail.

• View recent communication activities so that you can respond faster. See who called you and when. View voice messages onscreen and click to play or return the call. Message counters tell you how many voicemails and missed calls are waiting.

• Add communication media on demand. When on a call, you can quickly and easily add video or Web conferencing 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 send the call to voicemail with a simple mouse click.

Example of the Cisco Unified Personal Communicator with Video

Key Features and Benefits



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

• Presence: View real-time availability of other Cisco Unified Personal Communicator users. You can also choose to display customized messages, set out-of-office alerts, and show your availability based on your Microsoft Outlook calendar.

• 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 conversation; for example, you can add video to an audio conversation or add Web conferencing or whiteboarding to an existing audio or video conversation.

• Click to call: Dial from the contact list, using either the integrated softphone or an associated Cisco Unified IP phone. In addition, you can also click to call directly from Microsoft Outlook using the new Outlook toolbar.

• Integrated voice and video calling: Exchange ideas face-to-face with a coordinated video display on the PC screen and audio conversation with the softphone. 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 videophone.

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

• Instant messaging: Chat in real time using instant messaging to save time and reduce phone tag.

• Conferencing: Create voice or videoconferencing 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® or Cisco Unity Connection voicemail messages-view, playback, sort, and delete messages-all from within the application.

4.4 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:

Number of calls in queue;



Call appearance status; Calling/called party number/name; Trunk ID; COS/COR; Number of calls waiting; Call coverage status; Time/date; Call duration; Text messages; System alarm notification

Vendor Response Requirement Confirm the proposed attendant position 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 description technical PC requirements necessary to operate the soft client package. Also provide as an attachment 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 proposed the Arc Enterprise Attendant Console to meet this requirement. Arc Enterprise is an attendant console application for Cisco Unified Communications Manager that offers an advanced set of telephony functions combined with presence management and directory features.

Arc Enterprise offers a scalable, modular solution to organizations wishing to enhance customer service in their call centers by having a specially trained and equipped operator handle each inbound call.

Arc Enterprise: Key Features Three configurable directories (SQL)

Microsoft Active Directory support

User-definable search fields

Transfer and transfer recall

Hold and hold with notes

Camp-on, camp-on recall, and indication

Call park and call park recall

Leave a message by e-mail

Call toggle (call brokering)

Console speed dials

Time of day routing

Emergence queue feature

Whisper page to Cisco Unified IP phones, XML capable

Group page to Cisco Unified IP phones, XML capable

Presence management feature*

Personal call park feature*

Voice connect: queue messaging*

Server resilience options*

Custom keyboard*

Call statistics, graphs, reports*



Screen shot of Arc Attendant Console. Complete the following table for attendant soft console display requirements: Display Characteristic Yes/No

Number of calls in queue Yes Call appearance status Yes Calling/called party number/name Yes Trunk ID Yes COS/COR Yes Number of calls waiting Yes Call coverage status Yes Time/date Yes Call duration Yes Text messages Yes System alarm notification No

4.5 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 Confirm the proposed solution supports an IP audio conferencing unit and provide a brief description. Include as an attachment an illustration or photograph (PPT format, only) of the unit.

Cisco Unified IP Conference Station 7937G (shown with optional external microphones)

Cisco Response: The Cisco Unified IP Conference Station 7937G has been included to meet this requirement. The Cisco Unified IP Conference Station 7937G combines state-of-the-art wideband speakerphone conferencing technologies with award-winning Cisco voice communication technologies. The net result is a conference room phone that offers superior wideband voice and microphone quality, with simplified wiring and administrative cost benefits. A full-featured, IP-based, hands-free conference station, the new Cisco Unified IP Conference Station 7937G is designed for use on desktops, in conference rooms, and in executive suites.

The Cisco Unified IP Conference Station 7937G offers many improvements over the existing Cisco Unified IP Conference Station 7936. New features include:

Superior wideband acoustics with the support of the G.722 wideband code

Support for IEEE Power over Ethernet (PoE) or the Cisco Power Cube

Expanded room coverage up to 30 feet by 40 feet with the optional external microphone kit

Support for a third-party lapel microphone kit

New larger backlit liquid crystal display (LCD)

Global localization within six months of first customer shipment (FCS)



Features and Benefits

Superior wideband acoustics - Wideband frequency response is more than double that of narrowband, resulting in a richer and clearer sound. Enhances speech quality and speaker recognition, and provides excellent voice quality for both wideband and narrowband calls.

Expanded room coverage - Extends room coverage up to 30 feet by 40 feet with the optional Cisco Unified IP Conference Station 7937G microphone kit.

Support for third-party wireless lapel microphone kit - Allows presenters to be easily heard even if walking away from the Cisco Unified IP Conference Station 7937G microphones.

Large backlit LCD - Increases visibility even in dark areas (255 x 128 pixels).

Enhanced dial keys - Makes dialing easier.

Four soft keys - Quickens access to features.

IEEE 802.3af line power - Local power supply is not required.

Support for the Cisco Power Cube 3 - Uses standard Cisco Unified IP Phone Power Supply

Extensible Markup Language (XML) services - Support for Cisco XML services includes Cisco Unified Communications Manager and third-party directories, Cisco Web Dialer, Extension Mobility features, and Personal Assistant.

Support for Cisco Unified Communications Manager - Supports Cisco Unified Communications Manager Releases 4.1, 4.2, 4.3, 5.1, 6.x and 7.x.

4.6 Mobile Cellular Extensions VoiceCon requires that the proposed IPTS support mobile cellular extensions for a number of its station users. Two types of mobile cellular extensions are required: basic and advanced. Key characteristics of mobile cellular extensions include One Number Reach, User Programmable Ringing Sequence, Call Records, and Single Voicemail Box 4.6.1 Basic Mobile Cellular Extension Option VoiceCon requires that the proposed communications system solution support a basic mobile cellular extension option.

The option should be capable of working with almost any cellular carrier network and mobile handsets;

The mobile handset must be able to receive incoming calls directed to the station user’s primary system directory number, and calling party information should be displayed at the mobile handset;

Calls placed from the mobile handset to internal communications system subscribers must appear to look like calls from the station user’s primary desktop voice terminal, including calling party name/ID display;



The mobile handset must be able to place calls through the IPTS to external stations users;

IPTS system subscribers must be able to program incoming calls to ring simultaneously or sequentially at the desktop instrument and mobile handset as required;

Basic IPTS telephony features that should be supported in mobile extension mode, including Hold, Transfer, Conference, and Forward to IPTS voice mail system on busy/no-answer;

Call detail records must be collected and stored for all mobile extension calls.

At time of system installation 200 mobile cellular extension user licenses (or equivalents) must be included in pricing proposal. Vendor Response Requirement Confirm that your proposed communications system solution supports, at minimum, each of the specific mobile cellular extension capabilities as listed above. Include a brief description of any hardware/software requirements, including peripheral application servers, necessary to support the option and a provide a list of standard feature/function capabilities.

Cisco Response: Cisco Unified Mobility is natively available with Cisco Unified Communications Manager 7.x. In this proposal, mobility has been provided for all users (See Cisco Unified Workspace Licensing explanation in Section 1.2.0 above). Cisco Unified Mobility extends rich call control capabilities of Cisco Unified Communications Manager from a mobile worker’s primary workplace desk phone to any location or device of their choosing. Cisco Unified Mobility intelligently manages, filters, routes, and places calls between a worker's IP phone and remote mobile phone. Key Cisco Unified Mobility Features:

Single Business Number Reach & Single Business Voicemail - Cisco Unified Mobility makes it possible for workers to consolidate all their incoming business calls (i.e. incoming business calls to mobile phones, home office phone, or any temporary telework phone) into a single business phone number and immediately receive them wherever they are working. If mobile workers are unable to answer the call extended to one of the many user-defined alternative phone numbers, they can rely on Cisco Unified Mobility to store the unanswered calls into a single business voicemail on Cisco Unity or other business voicemail system.

Seamless Transition of Ongoing Extended Communications - Users can start the conversation from their mobile phone and seamlessly transition that conversation to a desk phone upon arrival in the office without needing to call back. Similarly they can transition the call seamlessly back to the mobile phone and wander away at will for another appointment.

Cisco Mobile Voice Access - With Cisco Unified Mobility, workers who



need to place national or international calls from their mobile phone can use the Cisco Mobile Voice Access line to place the call as if they are placing the call from their business extension on their desk. The worker dials the Cisco Mobile Voice Access line from the mobile phone and places the call on the IP communications network over a tie line. The call is connected and remains in control of Cisco Unified Communications Manager providing the opportunity to reduce mobile communications costs associated with national or international calls placed directly from mobile phone.

Access to Mid Call Control Cisco Unified Communications Manager Capabilities - Cisco Unified Mobility extends key call control features of Cisco Unified Communications Manager (such as hold, resume, transfer, and conferencing) on calls extended to devices and locations of the workers choice.

Personalized Access Lists - Mobile workers can access the secure user profile webpage to enter mobile and other alternate phone numbers and create filters that restrict the types of calls that are extended using Cisco Unified Mobility. Cisco Unified Mobility intelligently manages, filters, and routes each call between a worker's business extension and alternate phone numbers based on rules defined by the worker on their profile. Unanswered calls are consolidated into single business voicemail and voice communications resources are only used to extend relevant calls as determined by rules specified by the worker.

Web-Based System Administration - Cisco Unified Mobility provides system administrators with the flexibility to define and manage user profiles. System administrators can use the secure Administration Webpage 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 system administrator retains control over resource use and can provide backup support.

System Administration - In Cisco Unified Mobility Release 1.2, the system administration tools have been updated to include the Disaster Recovery System (DRS), which is also used in Cisco Unified Communications Manager Release 5.0. The Disaster Recovery System allows system administrators to back up and restore configuration and user data for software application upgrades and system failures. The Disaster Recovery System utility allows both manual and automatic backup and restores.

Security - In Cisco Unified Mobility Release 1.2, system security has been updated to support the Cisco Security Agent for Unified Communications Manager provides threat protection for Cisco Unified Communications application servers running voice applications, such as Cisco Unified Communications Manager, Cisco Unity Unified Messaging, and Cisco Unified Contact Center Express. Cisco Security Agent for Unified Communications Manager aggregates multiple security functions, combining host intrusion prevention, distributed firewall, malicious mobile code protection, operating system integrity assurance, and audit log consolidation, all within a single agent package. As part of an overall security strategy, the Cisco Security Agent for Unified Communications Manager enhances the SAFE Blueprint from Cisco and extends protection to the endpoint.



Cisco Unified Mobility 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 user's 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 Cisco Mobile Connect calls: During an active Cisco Mobile Connect call, the associated desktop 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 mobile phone.

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

• Single enterprise voice mailbox: Users can rely on their enterprise voicemail box as the single, consolidated voicemail box for all calls, 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 voicemail systems.

• Allowed and blocked call filters: Users can create a restricted list of caller phone numbers for which they want to trigger simultaneous 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 setup 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 Cisco 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 Unified Mobility Administration Webpages and by users through the secure User Profile Webpages. System administrators can determine how much control users have over their profiles, thereby preserving the administrator's ability to balance IP telephony resources with user choice.

• Remote on/off control: Users can turn Cisco Mobile Connect features on or off from a mobile phone using the Cisco Mobile Voice Access application or from the User Profile Webpages, 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 username and password.

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



4.6.2 Advanced Mobile Client To optimize the advantages of deploying the mobile cellular extension option VoiceCon also requires that the proposed communications system solution support an advanced mobile client option. The mobile client should include a user friendly GUI to facilitate and enhance mobile handset telephony service features and functions. At minimum the mobile client GUI should be capable of displaying: multi-line appearances; fixed and programmable feature keys; contact directories; call logs; incoming call identification information; internal call diversion information; active call information. All features/functions supported by the basic mobile extension option (see above) must also be supported by the advanced mobile client option. At time of system installation VoiceCon requires 100 advanced mobile client user licenses included in the pricing proposal. Vendor Response Requirement Confirm that your proposed communications system solution supports an advanced mobile client option and that it satisfies the capabilities listed above (specify capabilities not satisfied).

Provide a brief description of the mobile client’s general capabilities and features/functions;

List the cellular handset models and operating systems capable of supporting your mobile client option;

Provide as an attachment one or two graphical illustrations (PPT format, only) that are representative of the GUI screen display.

Cisco Unified Mobile Communicator (shown on a Nokia E61 phone)



Cisco Response: Cisco has included the Cisco Unified Mobile Communicator to meet this requirement. In this proposal, Cisco Unified Mobile Communicator licenses have been provided for all users. (See Cisco Unified Workspace Licensing explanation in Section 1.2.0 above) Cisco Unified Mobile Communicator is an easy-to-use software application for mobile handsets that extends enterprise communications applications and services to mobile phones and smartphones. It streamlines the communication experience, enabling real-time collaboration across the enterprise. Cisco Unified Mobile Communicator is currently supported on:

Blackberry OS: Blackberry (8300, 8310, 8700g, 8700v, 8703, 8800, 8820 and Pearl (8100))

Symbian OS: Nokia E61, E61i and E65 devices

Windows Mobile OS: HTC S620, Motorola Q9H, Motorola Q9C, Samsung Blackjack II, Samsung ACE, Toshiba Protégé G710

With Cisco Unified Mobile Communicator, users can:

Place and receive calls, click to dial from contacts list

Access company directory contacts

Access personal contact list

Check presence information to view real-time availability of other Cisco Unified Mobile Communicator users

Review Cisco Unity voice mail messages, playback in any order, and delete

Receive Cisco Unified MeetingPlace notifications and other vital information from Cisco Unified Communications

Send and receive secure text messages

Use integrated call logs to review lists of missed, received and placed calls

Two web-based management interfaces are provided with Cisco Unified Mobile Communicator:

Administrative Portal – for configuration and administration, setting up groups and priviledges, setting up security options, defining system parameters and collecting statistics.

End-User Portal – for phone provisioning and configuration, personal phonebook management, calendar integration, sending notifications and setting individual preferences.

The following screen shots illustrate examples of these Cisco Unified Mobile Communicator features:



Directory entries display real-time presence information and users can click-to-call from here.

Cisco Unified Mobile Communicator users can access voicemail from their mobile device and play them back in any order desired.

4.7 Other IP Telephone Instruments Include as an attachment a graphical illustration (PPT format, only) of IP telephone instrument models and add-on options not included as part of the proposed required system configuration.

Cisco Response: A PowerPoint file has been attached showing the additional IP phones. Also, a short Appendix C has been included which describes the features of these phones.



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 Unified IP Communications System. For the latest information please refer to Cisco Unified Communications Systems documentation posted at: www.cisco.com A comprehensive listing of Cisco Unified Communications Manager Release 7.0 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

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

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

USER CONTROLLED MEET ME CONFERENCING (6-Party or more) Yes

MESSAGE WAITING ACTIVATION Yes

MULTI-PARTY ASSISTED CONFERENCE w/SELECTIVE CALL DROP

MUSIC ON HOLD

Yes

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

Yes

STEP CALL Yes

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 IPTS 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. Also identify optional hardware/software, e.g., CTI application server, to satisfy a listed feature, because it is 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.

5.1.1 Additional Station User Features Vendor Response Requirement Provide as Attachment A a listing of all standard generic software station user features included with your IPTS solution.

Cisco Response: A detailed list of Cisco Unified Communications Manager station features is included in Appendix A.

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

Cisco Response AUTO-MANUAL SPLITTING

Yes AUTO-START/DON'T SPLIT

Yes BACK-UP ALERTING

Yes BUSY VERIFICATION OF TERMINALS/TRUNKS

No CALL WAITING

Yes CAMP-ON

Yes 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

Yes PAGING/CODE CALL ACCESS

Yes PRIORITY QUEUE

Yes RECALL

Yes RELEASE LOOP OPERATION

Yes SERIAL OPERATION

yes STRAIGHT FORWARD OUTWARD COMPLETION

Yes THROUGH DIALING

Yes TRUNK-TO-TRUNK TRANSFER

Yes TRUNK GROUP BUSY/WARNING INDICATOR

No TRUNK ID

Yes Vendor Response Requirement Confirm that the proposed IPTS supports each of the above listed attendant features. Identify any and all features that are not included as part of the standard call processing software generic package. Also identify optional



hardware/software, e.g., CTI application server, to satisfy a listed feature, because it is not included as part of the standard generic software package.

Cisco Response: Cisco supported features on the Arc Attendant Console application 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.

5.2.1 Additional Attendant Operator Features Vendor Response Requirement Provide as Attachment B a listing of all standard generic software attendant operation features included with your IPTS solution.

Cisco Response: A detailed list of Cisco Arc Attendant Console features is included in Section 4.4 above.

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

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 Yes

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

FIND ME- FOLLOW ME

No

Yes

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 Yes

UNIFORM CALL DISTRIBUTION Yes

UNIFORM DIAL PLAN Yes

VIRTUAL EXTENSION Yes

VOICE MESSAGE SYSTEM INTERFACE Yes

Vendor Response Requirement Confirm that the proposed IPTS supports each of the above listed system features. Identify any and all features that are not included as part of the standard call processing software generic package. Also identify optional hardware/software, e.g., CTI application server, to satisfy a listed feature, because it is 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.

5.3.1 Additional System Features Vendor Response Requirement Provide as Attachment C a listing of all standard generic software system features included with your IPTS solution.

Cisco Response: A detailed list of Cisco Unified Communications Manager 7.0 system features is included in Appendix A.



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 Communications Manager is administered via a web browser. All Cisco Unified Communications Manager 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. The servicibility tools, reports and plug-ins described below such as, RTMT, are included at no charge with Cisco Unified Communications Manager, thus the Management section of the pricing summary shows zero (0) dollars.



Any PC on the WAN/LAN with appropriate security can gain access to Unified Communications Manager 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). 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 Communications Manager 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 Communications Manager 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 Communications Manager 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 Communications Manager 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 Communications Manager SNMP trap messages that can be sent to an NMS that is specified as a trap receiver:

Cisco Unified Communications Manager 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 Communications Manager will send through the “Alarm Configuration” interface.

6.0.4 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. Vendor Response Requirement Confirm that the proposed management system supports user interface and tools as described above.

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. 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. 6.2 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 Communications Manager 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 Communications Manager services for troubleshooting and provides alarm message definitions.

Trace--Saves trace information generated by Cisco Unified Communications Manager 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 Communications Manager cluster.

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

Control Center--Displays status of Cisco Unified Communications Manager 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 Communications Manager Administration window by choosing “Applications” from the menu bar. Installing the Cisco Unified Communications Manager software automatically installs Serviceability and makes it available.

The Cisco Unified Communications Manager 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 fully 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 any of the listed attendant parameters which are not measured.

Cisco Response: Cisco does not report the above information in an Attendant Console specific manner. Information about call handling for Attendant Console clients will be obtained via the station and pilot point utilization

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 any of the listed 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 Communications Manager network and ended in the same Cisco Unified Communications Manager 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 Communications Manager network going out through the PSTN.

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

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

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

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

Details the 5 longest calls for the date range specified.

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, if applicable to your IPTS solution. 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. Identify what is not measured and reported.

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 of the listed items.

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 Communications Manager 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 of the listed devices. Identify what is not tracked, measured and reported.

Cisco Response: Cisco complies. Administrators use Alarms to obtain the runtime status and state of the Cisco Unified Communications Manager 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. Provide a listing of tracked features.

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, 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.

Cisco Response: Cisco complies 6.2.10.1 VoIP Call Recording The proposed management system should track and record the following VoIP call parameters:

CODEC Frame size Packets per frame Call type State of echo canceller State of silence suppression Total packets sent Total packets received Total packets late Total packets early Current jitter buffer level Maximum jitter buffer level Minimum jitter buffer level Packets discarded Silence frames inserted Effective packet loss Service state of each port

Vendor Response Requirement Confirm that the proposed management system tracks and records each of the



above listed VoIP call parameters. Identify any parameters not tracked and recorded.

Cisco Response: Cisco reports the following information in each Call Detail Record: call type, packets received, packets sent and packets lost. Cisco also records additional call details in the Cisco Maintenance Records (CMR). CMR contains codec, frame size, packets per frame, max jitter, packets discarded, effective packet loss. Information such as Codec, frame size, packets per frame, packets sent/received, current jitter, maximum jitter, packets discarded and packet loss can also be seen realtime on the phones. Cisco does not report any information about echo cancellation, silence suppression, silence frames inserted, or service state of each port.

6.2.10.2 VoIP Media Gateway Monitoring and Recording The proposed management system should track and record the following VoIP media gateway parameters:

Codec Frame size Number of active calls Total Calls Total Successful calls Total Connected calls Total RNA calls Total user Busy calls. Top ten termination reasons Average call hold time Maximum call hold time Average BHCA Maximum BHCA Top 10 busy minutes Number of Call signaling messages of type (x) received Number of Call signaling messages of type (x) sent Average response time for call signaling message type (x) Current active calls, including ANI, DNIS, call start time, call type,

current connect time, etc Vendor Response Requirement Confirm that the proposed management system tracks and records each of the above listed VoIP media gateway parameters. Identify any parameters not tracked and recorded.

Cisco Response: Via the CDR Analysis and Reporting (CAR), information regarding Codec, frame size. BHCA average and total placed call. But this information is only returned from MGCP controlled gateways. H323 gateways do not return this information. Live time information on the gateway’s current



status, active calls, total calls, successful calls and connected calls can be viewed live time via the Real Time Monitoring Tool (RTMT). If further call information is required, the administrator can access the IOS GW directly and view current call information immediately. At the present time, Cisco does not provide information related to hold times or signal messages send or received.

6.2.10.3 VoIP Transmission Reports The proposed management system should track and record the following VoIP transmission parameters:

Timestamp Information - start and stop time for the metric

measurement interval Stream Identification Information - differentiating between multiple

streams in which a given endpoint may be involved at a point in time Source - IP address, RTP port, and SSRC for monitored transmission

stream Destination - destination IP address, RTP port, and SSRC for monitored

transmission stream Codec Information Line – info for vocoder used for session Sample Rate - rate, in kiloHertz (kHz), at which the source audio is

sampled Frame Size - RTP packet size (bytes of the RTP frame) Packet Loss Concealment - indicator of whether packet loss concealment

is used Vendor Response Requirement Confirm that the proposed management system tracks and records each of the above listed VoIP transmission parameters. Identify any parameters not tracked and recorded.

Cisco Response: Cisco reports the following information in each Call Detail Record: call type, packets received, packets sent and packets lost. Cisco also records additional call details in the Cisco Maintenance Records (CMR). CMR contains codec, frame size, packets per frame, max jitter, packets discarded, effective packet loss. Information such as Codec, frame size, packets per frame, packets sent/received, current jitter, maximum jitter, packets discarded and packet loss can also be seen realtime on the phones. Cisco does not report any information about echo cancellation, silence suppression, silence frames inserted, or service state of each port.

6.2.10.4 VoIP Jitter Buffer Reports The proposed management system should track and record the following VoIP jitter buffer parameters:



Jitter Buffer Information Line – info re jitter buffer within the media

endpoint. Jitter Buffer Type – indication if jitter buffer is adaptive or static. Jitter Buffer Adaptation Rate - specific adjustment rate of a jitter

buffer in adaptive mode. Jitter Buffer Nomimal Delay - current nominal jitter buffer delay in

milliseconds, which corresponds to the nominal jitter buffer delay for packets that arrive exactly on time.

Jitter Buffer Maximum Delay - current maximum jitter buffer delay in milliseconds corresponding to the earliest arriving packet that would not be discarded

Jitter Buffer Absolute Maximum Delay - absolute maximum delay in milliseconds that the adaptive jitter buffer can reach under worst case conditions

Vendor Response Requirement Confirm that the proposed management system tracks and records each of the above listed VoIP jitter buffer parameters. Identify any parameters not tracked and recorded.

Cisco Response: The Cisco Management application will report information about jitter, but not all of the above. All of Cisco’s products support adaptive jitter buffers (with the one exception being the MTP and SW based conferencing). As such, the only jitter values that will be reported is the max jitter value. For Cisco IOS gateways, one can obtain and set additional playout values on the gateway directly, but this information will not be reported back to the RTMT management station.

6.2.10.5 VoIP Packet Reports The proposed management system should track and record the following VoIP packet parameters:

Packet Loss Information Line - general packet loss metrics Packet Loss Ratio - percentage of packets lost within the network

during the time period captured by the report Packet Discard Rate - percentage of packets discarded due to jitter

within the network during the time period captured by the report Burst Loss Information Line - parameters in this provide burst loss

metrics Burst Density - percentage of packets lost and discarded within a

burst (high loss rate) period. Burst Length (mS) - mean length of a burst. Gap Loss Information Line - parameters in this provide random loss

metrics



Vendor Response Requirement Confirm that the proposed management system tracks and records each of the above listed VoIP packet parameters. Identify any parameters not tracked and recorded.

Cisco Response: Cisco Unified Real Time Report Tool (RTMT) does not include all of these parameters in it’s reports. The Cisco Unified IP Phones does track Lost Packets, but when the information is recorded in the CMR records, it included both packet lost and discards as the same. Most of these measurements are too granular and will not provide an administrator with much value. The IP endpoint should use these values to adjust playout, but if there is no action that can be taken to correct the individual parameters, then the value of overall packet loss will far exceed any other packet loss reports that are returned.

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.

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. Confirm the proposed management system compiles data for each of the above CDR parameters. Identify any listed CDR parameters not tracked and recorded.

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 Communications Manager CDR format. There is a searchable database at the following url that shows the current list of partners. It also allows you to search for information about partners in other application areas:

http://www.cisco-partners.info/ 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 Communications Manager SQL data. Call Detail Reporting (CDR) is a standard feature of Unified Communications Manager. Cisco Unified Communications Manager 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 Communications



Manager. CDR and CMR options are configured in Cisco Unified Communications Manager 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 Communications Manager 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 Communications Manager. 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 Communications Managers in a cluster is collected on one designated Unified Communications Manager server, rather than being collected separately in each of them. 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 Communications Manager 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 for the required maintenance monitoring activities by



completing following table: Activity Yes/No Monitoring of processor status Yes Monitoring and testing of all port and service circuit packs Yes Monitoring and control of power units, fans, and environmental sensors

Yes

Monitoring of peripherals (voice terminals and trunk circuits) Yes Initiate emergency transfer and control to backup systems Yes Originate alarm information and activate alarms Yes 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 Provide a list all standard 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 Communications Manager 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 Communications Manager 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 Communications Manager configuration data and system information. Cisco Unified Communications Manager supports the following capabilities for remote serviceability:

• Cisco Secure Telnet—Allows CSEs to log on to customer remote site to troubleshoot Cisco Unified Communications Manager system.

• Command Line Interface—Allows CSEs to display Cisco Unified Communications Manager system statistics and troubleshoot the Unified Communications Manager 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 Communications Manager 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 Communications Manager servers and management of those servers by CiscoWorks2000.

Cisco Secure Telnet Cisco Secure Telnet offers Cisco Service Engineers transparent firewall access to Cisco Unified Communications Manager 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 Communications Manager 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: Cisco Unified Communications Manager is administered via a web browser. All Cisco Unified Communications Manager 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. In general, each application is managed directly rather than through a consolidated interface. The servicibility tools, reports and plug-ins described above such as, RTMT, are included at no charge with Cisco Unified Communications Manager, thus the Management section of the pricing summary shows zero (0) dollars. The entire cluster is managed 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 Communications Manager clusters in an Enterprise, each cluster must be managed separately. There is no single web page to manage multiple clusters. Additional Tools Available from Cisco: This RFP does not specify the underlying converged IP infrastructure, so no additional or optional network management tools were included in this proposal. However, VoiceCon may be interested in these optional Network Management tools available from Cisco. These management tools are described in detail at the following url: http://www.cisco.com/en/US/products/sw/netmgtsw/index.html The network management tools specifically designed for unified



communications solutions include the following, briefly described below: http://www.cisco.com/en/US/products/ps5747/Products_Sub_Category_Home.html

For enterprises with 1000 to 30,000 devices, the Cisco Unified Communications Management Suite is an integrated set of products designed to provide a single view into the entire Cisco Unified Communications solution to help you increase operational efficiency. The suite consists of:

Cisco Unified Provisioning Manager Cisco Unified Operations Manager Cisco Unified Service Monitor Cisco Unified Service Statistics Manager



7.0.0 Integrated Messaging System VoiceCon requires a CHQ-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 CHQ location, and be capable of supporting station users at the twol remote VoiceCon faciliities (RO and SO). 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:

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 or IBM Lotus Domino system. Include processing system platform information in the discussion. Verify that the system being bid can comply with each of the proceeding requirements.

Cisco Response: Cisco is proposing Cisco Unity Release 7.0 for Microsoft Exchange to meet the unified messaging and automated attendant requirements. This system is installed at the headquarters location. Cisco Unity integrates to Unified Communications Manager through a SCCP/TAPI or a SIP connection. Unity Voice Mail is available in various session configurations supporting up to 200 ports and 15,000 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 96-session configuration installed as a turnkey system on one Cisco MCS 7845-H2-ECS2 server. This turnkey solution comes completely assembled and tested with all the software installed. This configuration provides 96 ports installed

Installed/Equipped Capacity Maximum Capacity Number of Subscribers 2000 3,000 Number of Ports 96 128 Hours of Storage 1000 1500



capacity for growth. Cisco Unity Unified Messaging integrates transparently with Microsoft Exchange, allowing you to handle all your messages -- e-mail, voice, and fax -- through a single inbox using the Outlook e-mail client. 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 also can reply to, forward, and save your messages -- regardless of media type -- in public or personal Microsoft Outlook folders with just a click of the mouse, decreasing response times and increasing organizational agility and customer service. With the Text-to-Speech (TTS) capability of Cisco Unity Unified Messaging, you get information about all your messages -- and even hear the text portion of e-mail messages -- over the telephone. Depending on the capabilities of your fax server, you can even print e-mail, attachments, and incoming faxes on a nearby fax machine. Even when away from your computer, you are never away from your vital business communications. Cisco Unity Features – A complete list of Unity for Microsoft Exchange 7.0 features and capabilities is included in Appendix B.

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

5.0(1) Windows Vista XP 2000

5.0(2) 5.0(1)

Cisco Unified Personal Communicator

1.2(1) Outlook

2007 2003 2002

(XP) 2000

Internet Explorer

7.0 and later

6.0 (plus Service Pack 1)

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



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



5.0(1) Windows XP 2000

1.2.3 1.2.2

Lotus Domino Web Access

7.0 Lotus iNotes

6.5.x Lotus Notes

7.0 6.5.x

Internet Explorer 6.0 (Plus

Service Pack 1) 5.5 (Plus

Service Pack 2)

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:

Compatibility Matrix: Cisco Unity and the Software on Subscriber Workstations [Cisco Unity] - Cisco Systems

7.1.1 Security Features Vendor Response Requirement Describe all 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 Communications Manager and Unified IP Phones.

Encryption: Cisco Unity Unified Messaging can encrypt messages as they are taken. You can then listen to the messages either through the telephone user interface (TUI) or from Outlook and the messages will be properly decrypted. With Secure Messaging, if a message is then forwarded outside the organization, the recipient of the forward will be unable to decrypt that message. Thus you can use Secure Messaging to prevent messages from leaving the organization.

Private Messages: Additionally, you can mark messages as private, in which case the encryption mechanism limits playback to only the original intended recipient. You can also configure encryption keys to expire after a set period of time, meaning messages will become unplayable records after the end of the expiry period even if copied to a computer hard drive. With Secure Messaging you can be assured that your security and compliance policies will be strictly adhered to while still allowing users the benefits of unified



messaging. 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 Communications Manager and Unified IP Phones Cisco Unity supports security features to Unified Communications Manager 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 Communications Manager 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:

All other mailbox owners, and 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 must 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:

Storage space used for announcements or information mailboxes Storage space used for messages Maximum storage space used during the interval

Vendor Response Requirement Confirm support for this feature. Indicate if any requirement is not supported. 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:

Storage space used for announcements Total calls answered Total calls routed to station Total calls routed to default Total calls abandoned 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.

Cisco Response: Cisco Unity offers several networking options:

Cisco Unity Digital Networking: Cisco Unity offers an optional digital networking module that allows the system to connect to other Cisco Unity servers at the same site through the LAN, or remote sites using a WAN or the Internet. Digital networking makes communicating with co-workers at remote locations fast and efficient by allowing you to send subscriber-to-subscriber messages anywhere in the world. With digital networking you can use the Global Addressing feature -- listing all system subscribers in a central directory -- to quickly and conveniently send a message to a co-worker in another time zone. Subscriber-to-subscriber messages offer more reply options to the recipient, making it simpler to respond to an e-mail with a voice message, for example. Also, when retrieving messages over the telephone, voicemail from system subscribers is played with the sender's recorded name for greater recognition. Cisco Unity Bridge: A powerful message networking option available with Cisco Unity is the Cisco Unity Bridge. With Cisco Unity Bridge, you can send subscriber-to-subscriber messages to anyone in your organization who resides on a TDM-based Avaya or Octel voicemail system supporting Octel Analog Networking. In addition, you can simply "reply to" a networked message with a single touch-tone key. With Cisco Unity Bridge, you can maintain advanced messaging capabilities on both systems as you migrate to the Cisco Unity system.



AMIS and VPIM: Cisco Unity also provides optional Voice Profile for Internet Mail (VPIM [digital]) and Audio Messaging Interchange Specification (AMIS [analog]) networking modules that allow message interchange between disparate messaging systems that also support these industry-standard messaging protocols. With VPIM or AMIS, customers who are migrating to Cisco Unity can continue to exchange messages with internal system subscribers who reside on a third-party messaging system, helping to ensure a smooth system migration.

7.3.1 AMIS The proposed messaging system must support AMIS networking standards. Vendor Response Requirement Confirm support for this networking capability.

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 is optional and 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 VoiceCon requires that its voice messaging system be fully integrated with a Microsoft Exchange or IBM Lotus Domino messaging solution to provide unified messaging access from any system termnial/client. Vendor Response Requirement Confirm that the proposed voice messaging system supports this requirement. Briefly describe how the proposed voice messaging system can be integrated with VoiceCon ’s text messaging system, based on both Exchange and Notes servers running the latest software versions 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.



Section 8: Unified Communications

8.0 Unified Communications

VoiceCon may decide to a departmental trial of a Unified Communications (UC) solution prior to implementing it across the entire organization. VoiceCon defines a UC solution as one that supports the following features and capabilities:

Station user programmed call screening, coverage, and routing Presence management & control Instant Messaging (IM) Conferencing tools for audio, data, and video communications Collaboration tools for desktop screen sharing, management and editing

Vendor Response Requirement:

Confirm your proposed IPTS communications solution can support an integrated UC option that satisfies the general requirements and capabilities outlined in the following RFP sections, and provide a brief overview of your UC offer. 8.0.1 UC System Integration

Vendor Response Requirements: How can the proposed IPTS and voice messaging systems physically and logically integrate with the available UC solution?

Cisco Response: Cisco has proposed a complete Unified Communications system including voice, video, collaboration, unified messaging, mobility, presence, security and administration and maintenance. These capabilities have been provided for all users on the system, not just a subset. Cisco’s Unified Workspace Licensing program makes this possible at a very cost-effective price. VoiceCon will be able to enhance business productivity by creating a Unified Workspaces encompassing multiple applications, devices, networks, and operating systems. VoiceCon will be able to improve integration of communications with business processes to ensure that information reaches recipients every time, regardless of their working environment or location.

Cisco Unified Communications Solutions include network-powered solutions for:

IP telephony: Cisco IP Telephony allows companies to realize the benefits of Voice over IP (VoIP) technology with a suite of media control and IP phone solutions.

Rich media conferencing: Cisco Unified MeetingPlace provides a complete multimedia conferencing solution for meetings, training sessions, and presentations.



Unified messaging: Cisco Unified Messaging integrates all types of messages so they can be easily accessed from a single user interface regardless of location or device.

Unified communications clients: Cisco Unified Personal Communicator offers voice, video, instant messaging, conferencing, and presence information through a single application on a PC or Mac.

Mobility solutions: Cisco Mobile Solutions for Unified Communications combine the convenience, flexibility, and reach of mobile communications with the benefits of Cisco Unified Communications, making it possible for people to communicate the way they want, whether at home, at work, or on the road.

Collaborative work spaces: WebEx and Cisco TelePresence bring people together in a way that's never been done before while reducing travel expenses and carbon footprint.

Contact center: Cisco Unified Contact Center solutions integrate databases and workflow applications with advanced contact center applications, making it possible to create a virtual contact center where customers can be matched to the most appropriate agents.

Cisco’s broad offerings in Unified Communications are described in great detail at: http://www.cisco.com/en/US/products/sw/voicesw/products_category_technologies_overview.html

8.02 Presence Management Vendor Response Requirement: Confirm that presence management is fully integrated into the available UC solution. Include a brief description of presence features and functions included with the offering. Indicate if presence management requires a dedicated server or gateway with middleware.

Cisco Response: Cisco Unified Presence is a standards-based platform that collects information about a user's availability and communications capabilities to provide unified user presence status and facilitate presence-enabled communications for Cisco Unified Communications and critical business applications. With this scalable and easy-to-manage solution, Cisco Unified Presence delivers a consistent presence-enabled communications experience across Cisco Unified Communications applications everywhere, every time, independent of user device, application, or workspace location. In addition, Cisco Unified Presence gives customers and partners the flexibility to presence-enable and streamline business communications by interoperating with critical business applications through open interfaces.

Product Overview

Cisco Unified Presence takes advantage of Session Initiation Protocol (SIP) technology to support new voice services in the 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 uses SIP for



Instant Messaging and Presence Leveraging Extensions (SIMPLE) technology to support instant messaging and presence. Cisco Unified Presence also supports presence-oriented Simple Object Access Protocol (SOAP) and Representational State Transfer (REST) interfaces, providing customers and application developers the ability to extend presence to web-based business applications. Cisco Unified Presence consists of a SIP presence and instant messaging engine and a SIP proxy function. The presence engine collects user presence information (such as busy, on the phone, idle, away, or available status) as well as user capabilities (such as the capability to support voice, video, instant messaging, 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. Users can apply unique user rules and privacy to help ensure that only authorized applications and users have access to presence information. The SIP proxy function facilitates efficient and accurate routing of both presence and general SIP messaging through the enterprise. Cisco Unified Presence is tightly integrated with various desktop clients and applications. It helps Cisco Unified Personal Communicator, the Cisco Unified Communications enterprise desktop client, to perform numerous functions such as instant messaging, click to dial, phone control, voice, video, and web collaboration. In addition, Cisco Unified Presence provides a unified presence service for Cisco Unified Mobile Communicator, instant messaging service for Cisco Unified IP Phones connected to Cisco Unified Communications Manager, as well as providing the ability to integrate Cisco Unified IP Phone presence with IBM Lotus Sametime. In a separate mode of operation, Cisco Unified Presence also supports interoperability with Microsoft Office Communicator, allowing this desktop application to operate in conjunction with Cisco Unified IP Phones supported on Cisco Unified Communications Manager.

Cisco Unified Presence Modes of Operation

Cisco Unified Presence has two modes of operation: • Cisco Unified Communications mode: In this mode, Cisco Unified Presence operates as a SIP / SIMPLE presence server supporting Cisco Unified Communications clients such as Cisco Unified Personal Communicator and Cisco IP Phone Messenger. When operating in Cisco Unified Communications mode, Cisco Unified Presence provides presence, instant messaging, and click-to-call capabilities for Cisco Unified Personal Communicator from either a PC or MAC, and it scales up to a maximum of 30,000 users in a multi-node cluster environment.

• Microsoft Office Communicator Interoperability mode (or Microsoft Client mode): In this mode, Cisco Unified Presence provides Microsoft Office Communicator users on a PC ability to interoperate with Cisco Unified IP Phones on Cisco Unified Communications Manager by providing click-to-dial and associated phone-monitoring capabilities (refer to Figure 1). Interoperability is made available by activating Microsoft Client mode in Cisco Unified Presence and configuring Microsoft Office Communicator users. Cisco Unified Presence supports both Live Communication Server (LCS) 2005 and Office



Communication Server (OCS) 2007 in this mode. When operating in Microsoft Client mode, Cisco Unified Presence scales up to 10,000 Microsoft Office Communicator users per Cisco Unified Presence server and Cisco Unified Communications Manager cluster.

Cisco Unified Presence Maximum Capacities

Mode of Operation Maximum Users Supported

Cisco Unified Communications Mode

30,000

Microsoft Office Interoperability Mode

10,000*

* In this mode of operation Cisco Unified Presence operates as an interface point. Capacities for Microsoft Office Interoperability Mode are based on limits defined for a Cisco Unified Communication Manager cluster.

Cisco Unified Presence in Microsoft Office Communicator Interoperability Mode

Features and Benefits of Cisco Unified Presence

The following sections discuss the features and benefits of Cisco Unified Presence.

Enterprise-Class Scalability, Redundancy, and High Availability

Cisco Unified Presence provides enterprise-class scalability, redundancy, and high availability desired by large businesses and organizations. Full and partial highly available deployment models provide redundancy and resiliency to facilitate persistent client connectivity, presence aggregation and distribution when unplanned system outages occur. This also provides flexibility to schedule maintenance with minimal or no service disruption. In the full high-availability deployment models, each live Cisco Unified Presence server instance is mirrored in real time to a backup server ready to take control over the user clients. Alternatively, partial high-availability deployment models support Cisco



Unified Presence failover to a second live operational server that takes on the additional system load. By networking up to three Cisco Unified Communications Presence groups together, Cisco Unified Presence delivers scalability for up to 30,000 active Cisco Unified Presence users in Cisco Unified Communications mode.

Cisco Unified Presence Multinode Clustering

Standards-Based Application Programmable Interfaces

Cisco Unified Presence provides a standards-based connectivity to any SIP/SIMPLE capable application or network. In effect, you can request any user status that is maintained in the Cisco Unified Presence engine by using the IETF standards for status and presence sharing. These SIP/SIMPLE standards define the accepted messaging to initiate and maintain a status request and to provide appropriate responses. The presence engine can collect and distribute status information, depending on the needs of the services deployed. Cisco Unified Presence also supports web-centric application programmable interfaces (APIs) such as REST and SOAP. These APIs provide IT departments and system integrators with the simple but powerful ability to presence-enable their business applications; for example, exposing expert or user availability on a corporate web directory, point-of-sales application, or customer-relationship-management system.

Microsoft Outlook Calendar Integration

Cisco Unified Presence can incorporate Microsoft Outlook Calendar free and busy data when publishing a user's availability. This feature helps you maintain your availability and status information automatically and because it is based on a server-to-server integration, it is available to other users whether the originating user is logged in or not. The Microsoft Outlook Calendar feature



requires the establishment of a gateway connection to the Microsoft Exchange Server and works with both Microsoft Exchange Server 2003 and Microsoft Exchange Server 2007.

Cisco Unified Presence Federation

Federation is the secure, policy-controlled interconnection between different enterprises to enable them to securely exchange instant messaging and presence information between users in different businesses or organizations. In Cisco Unified Presence this capability is delivered between organizations that are both running Cisco Unified Presence 7.0 as well as when one organization is using Cisco Unified Presence 7.0 and the other is using Microsoft Live Communications Server (LCS) 2005 and Microsoft Office Communications Server (OCS) 2007. This capability delivers secure, policy-controlled Interdomain Federation between environments that have different domain names, such as [email protected] to [email protected], as illustrated in Figure 3.

Business-to-Business Federation

VMWare Player Support - Cisco Unified Presence supports VMWare for nonproduction environments such as demonstrations, lab systems, and development environments. VMWare player support aligns with VMWare player capability available in Cisco Unified Application Environment.

SIP Proxy Services for Cisco Customer Voice Portal - Cisco Unified Presence provides the SIP proxy services needed to support large Cisco Customer Voice Portal 4.0 (and higher) deployments. Centralizing the Cisco Customer Voice Portal dial plan using Cisco Unified Presence helps reduce initial setup time and ongoing administration of the Cisco Unified Communications Solution.

Platform Management, Security, and Support - Cisco Unified Presence uses the same platform infrastructure as Cisco Unified Communications Manager, following its appliance model principles. Cisco Unified Presence is a single software entity that provides access to administration with a GUI and allows



initial setup and installation through a command-line interface (CLI) similar to those for other Cisco products.

Security - The security module of Cisco Unified Presence addresses internal environment security as well as external security among Cisco Unified Communications Manager, Cisco Unified Personal Communicator, and external applications. Cisco Unified Presence supports:

• Distribution of engine and proxy trust certificates to other nodes of Cisco Unified Presence cluster through replication

• Adding trusted peer in SIP proxy Transport Layer Security (TLS)

• Automatic distribution of SIP proxy self-signed certificate or Certificate Authority root certificates

Serviceability - Cisco Unified Presence takes advantage of the same serviceability features as Cisco Unified Communications Manager. In addition, Cisco Unified Presence has enhanced capabilities in the areas of alarms, performance counters, debug and trace utilities, service activation, monitoring, and CLI.

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 ease movement of end-user in a Multi-node cluster environment

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

8.0.3 Instant Messaging Vendor Response Requirement: Confirm that Instant Messaging (IM) is fully integrated into the available UC solution. Include a brief description of IM features and capabilities, and also indicate if the integrated IM offer is interoperable with public IM services? Is IM recording supported?

Cisco Response: Chat: Users can send public and private text messages within MeetingPlace collaborative meetings. In presentation and Web seminar meetings, presenters can chat with anyone, and audience members can chat only with presenters. Moderated chat: Presenters and moderators can select specific questions to which to respond, either to the individual or to the entire group. Integrated IM: With Cisco Unified MeetingPlace Jabber Integration, users can easily initiate integrated voice, video, and Web conferences from Jabber Messenger. Similarly, with Cisco Unified MeetingPlace for Microsoft Office Communicator, users can initiate conferences directly from Microsoft Office Communicator. And with Cisco Unified Personal Communicator, users can



initiate Cisco Unified MeetingPlace Web conferences. 8.0.4 UC Clients Vendor Response Requirement: What types of UC-based desktop clients are supported by the available UC solution? Is a SIP-based client supported? Identify the PC’s technical specifications required to support the UC client. Provide as an attachment a graphical illustration (PPT format, only) of a typical UC screen shot that is representative of your UC offering.

Cisco Response: An integral component of the Cisco Unified Communications family of products, Cisco Unified Personal Communicator transparently integrates your most frequently used communications applications and services into a single, unified client. From an easy-to-use interface on a PC or Mac, it provides quick and easy access to powerful communications tools - softphone, presence, instant messaging, visual voicemail, click to call, employee directory, communication history, video, and web conferencing - to help you communicate effectively and work more productively.

Communicate More Effectively

Many workers battle communications overload daily, and they are forced to use a wide variety of devices and applications to communicate with colleagues, partners, and customers. Each of these applications works differently, with its own set of rules, tools, and directories. Cisco Unified Personal Communicator simplifies the communications experience by giving you quick and easy access to a unified set of communication tools. For example, using dynamic presence information and instant-messaging capabilities from Cisco Unified Personal Communicator, you can check the availability of colleagues and partners and chat 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 can help you exchange ideas "face-to-face" and collaborate more effectively with colleagues. You can also view and listen to voice messages quickly and easily. With Cisco Unified Personal Communicator, it is easy to access your communication and collaboration tools from every workspace, everywhere, every time for smarter, more effective communications.

Example of Cisco Unified Personal Communicator



Reduce Communication Delays with Colleagues, Partners, and Decision Makers

Cisco Unified Personal Communicator can help you determine if co-workers are available or busy before trying to contact them. Independent of whether co-workers are using Cisco Unified Personal Communicator on their computer, or a Cisco Unified IP Phone in their office, single aggregate contact availability information is updated automatically using dynamic information from Cisco Unified Presence. You can see immediately who is offline, available, away, on the phone, or in do-not-disturb state. Customized information, such as "on vacation" or "in a meeting", is also available to let you know why someone is unavailable or busy. Knowing whether contacts are available helps reduce communication delays between workers, thereby enabling faster decision making and enhanced productivity.

Streamline Communications

Cisco Unified Personal Communicator facilitates streamlined communications from your desktop or laptop computer, including integrated contact lists, click



to call, instant messaging, 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 be more productive: • Find contact information quickly by using Cisco Unified Personal

Communicator to search your corporate directory.

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

• Make calls using the integrated softphone while working away from the office or use Cisco Unified Personal Communicator to control your Cisco Unified IP Phone on Cisco Unified Communications Manager at your desk.

• Use the Cisco Unified Personal Communicator toolbar to click to dial from within your Microsoft Outlook contacts list or email.

• Exchange presence information and instant messages with business partners using Cisco Unified Personal Communicator or Microsoft Office Communicator when secure business-to-business federation is enabled between Cisco Unified Presence and Microsoft Live Communications or Microsoft Office Communications servers.

• View recent communication activities so that you can respond faster. See who called you and when. View voice messages onscreen and click to play or return the call. Message counters tell you how many voicemails and missed calls are waiting.

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

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

• Receive pop-up notifications of incoming calls with caller ID. You can accept the call if you are available or send the call to voicemail 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 can support your company's green initiatives, reduce travel expenses, and time associated with traveling to meetings. Using web conferencing, you can collaborate with co-workers virtually everywhere, every time. 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, instant messaging, voicemail playback, web conferencing, and integrated directories.



• Presence: View real-time availability of other Cisco Unified Personal Communicator and Cisco Unified IP Phone users. You can also display customized messages, set an out-of-office message, and automatically show your availability based on free and busy status on your Microsoft Outlook Calendar.

• Do not disturb (DND): Easily block incoming calls with synchronized DND status from your Cisco Unified Personal Communicator or Cisco Unified IP Phone or use the privacy preference setting to block instant messages when you need additional privacy.

• Contact list: Search your corporate directory from one easy-to-use interface to locate contacts quickly and simply click to call. Add your most frequently contacted personal contacts, co-workers, and federated business contacts

• Media escalation: Add communication methods during a conversation; for example, you can add video to an audio conversation or add web conferencing or whiteboarding to an existing audio or video conversation.

• Click to call: Dial from the contact list, using either the integrated softphone or an associated Cisco Unified IP Phone. You can also click to call directly from Microsoft Outlook using an Outlook toolbar.

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

• IP phone association: Use Cisco Unified Personal Communicator to control your desktop Cisco Unified IP Phone to make, receive, or merge calls.

• Instant messaging: Chat in real time using instant messaging with other Cisco Unified Personal Communicator users to save time and reduce phone tag. In addition, enable business-to-business federation between Cisco Unified Presence and Microsoft Live Communications or Microsoft Office Communications server to exchange presence information and instant messages with Microsoft Office Communicator and Cisco Unified Personal Communicator users.

• Conferencing: Create voice or video conferencing sessions by simply merging conversation sessions. There is no need to call into a separate conference bridge.

• Web conferencing: Launch a Cisco Unified MeetingPlace or Cisco Unified MeetingPlace Express web conferencing session at a moment's notice to share content, such as a presentation, with others.

• Voice messages: Access secure Cisco Unity® or Cisco Unity Connection encrypted voicemail messages - view, play back, sort, and delete messages - all from within the application.

System Requirements

The tables below give the computer requirements of Cisco Unified Personal Communicator for Microsoft Windows and Apple Macintosh, respectively.

Computer Requirements of Cisco Unified Personal Communicator for Microsoft Windows



Parameter Description

Disk space 200-MB free hard drive space (Includes 20 MB free space for the Windows camera drivers)

Hardware Microsoft Windows XP

• Desktop PC: • At least 2.4 GHz Intel Pentium 4, 2.0 GHz AMD Athlon (approximate speed) or faster processing (At least 2.8GHz Intel Pentium 4, 2.13 GHz AMD Athlon (approximate speed) or faster recommended for video calling capabilities) is required. In addition, 512MB of RAM for audio calls and 1GB RAM are required for video calls. • DirectX 9.0c-compatible graphics card with at least 64-MB free video RAM (64 MB for dual-headed cards) 1024 x 768 x 16 bits or better; for video calling, DirectX 9.0c-compatible graphics card with at least 64-MB free video RAM (128 MB for dual-headed configurations) • Laptop PC: • 1.5 Ghz Intel Pentium M Centrino 1.8 GHz AMD Athlon 2200 (Approximate speed) or compatible processing (1.7 GHz Pentium M (Centrino) 2.0 GHz AMD Athlon XP 2400+ (Approximate speed) or faster recommended for video calling capabilities • DirectX 9.0c-compatible graphics card with at least 64-MB free video RAM 1024 x 768 x 16 bits or better • Only 32-bit (x86) processors are supported. • A non-ISA full-duplex sound card (integrated or PCI-based) or USB sound device with USB headset is recommended when using a softphone. • A 10-/100/-1000BASE-T Mbps Ethernet network interface card is required. • A Cisco VT Camera II or third-party USB camera attached to a USB 2.0 port is required for video calls.

Microsoft Windows Vista

• A Microsoft Vista Premium Ready PC is required. For details about the minimum hardware requirements for Windows Vista (in addition to the requirements in this table), search for Premium Ready PC on the Microsoft website or refer to this URL: http://support.microsoft.com/kb/919183. • Only 32-bit (x86) processors are supported. In addition, 256MB of dedicated video memory is also needed • Hardware in computers running Vista and Cisco Unified Personal Communicator using video must have a base score of 3 or higher. Run the performance tool by choosing Start > Control Panel, and clicking Performance and Rating. • The subscores for memory (RAM), graphics, and gaming graphics must be 3 or higher. • Disk space: 200 MB of free disk space (includes 20 MB free space for the Windows camera drivers) is required. • A non-ISA full-duplex sound card (integrated or PCI-based) or USB sound device with USB headset is recommended when using a softphone. • A 10-/100-Mbps Ethernet network interface card is required. • A Cisco Unified Video Advantage Camera II attached to a USB 2.0 port is required for video calls.



Software • Microsoft Windows XP Professional (Service Pack 2 or Service Pack 3) or Microsoft Windows Vista Business Edition or Enterprise Edition - also supports Windows Vista Service Pack 1 (SP1) • (For Windows XP) Adobe Flash Player 6.0.79 or higher required for web conferencing • (For Windows Vista) Adobe Flash Player ActiveX or Adobe Flash Player 9 (Version 9.0.28 or higher) • Microsoft DirectX 9.0c • Microsoft Outlook 2003 or 2007 required for click-to-dial toolbar support

Connectivity High-speed connection required for softphone calls; 128 kbps for audio calls and 384kbps for calls with video

Computer Requirements of Cisco Unified Personal Communicator for Apple Mac OS X

Parameter Description

Disk space 200-MB free hard drive space

Hardware 1.4-GHz or faster PowerPC G4 or compatible processor; any Macintosh with PowerPC G5 or Intel processor recommended for video calling capabilities

A non-ISA full-duplex sound card (integrated or PCI-based) or USB sound device

A 10-/100-Mbps Ethernet network interface card

Memory 512-MB RAM (1-GB RAM recommended for video calling capabilities)

Software Mac OS X 10.4.11

Mac OS X 10.5.4

Macromedia Flash Player 6.0.79 or later required for web conferencing

Apple Address Book 4.0.4 (485.1) or later (available in Mac OS X 10.4.7) for local address-book support

Connectivity High-speed connection required for softphone calls; 128 kbps for audio calls and 384 kbps for calls with video

Note: A list of vendors that have verified their devices for use with Cisco Unified Personal Communicator through the Cisco Technology Developer Program is available at http://www.cisco.com/pcgi-bin/ctdp/Search.pl. These devices have passed lab testing and met interoperability criteria, ensuring that Cisco product specifications have been reached.

Minimum System Requirements

• Cisco Unified Communications Manager 5.1(3), 6.0, 6.1, or 7.0

• Cisco Unified Presence 6.0 or 7.0



• Cisco Unified IP Phones required for Deskphone Control mode (Note: Not all phone loads support computer telephony integration [CTI]; refer to Cisco Unified Communications Manager System Guide for more information)

• Cisco Unified MeetingPlace® Express 1.2 or Cisco Unified MeetingPlace 6.0 or 7.0 required for web conferencing features

• Cisco Unified MeetingPlace 6.0 or 7.0 required for whiteboarding features

• Cisco Unity Connection 2.0 or 7.0, or Cisco Unity 4.2, or 5.0 required for voicemail access

• Either Cisco Unified Videoconferencing 5.0 or 5.5 or Cisco Unified MeetingPlace Express VT 1.2 or 2.0 required for video conferencing

• Cisco ASA 5500 Series Adaptive Security Appliance Software Release 8.0.4.for business-to-business federation of presence and instant messaging between Cisco Unified Personal Communicator and Microsoft Office Communicator users

• Lightweight Directory Access Protocol Version 3 (LDAPv3) server

Note: Not all features are supported with all versions of system components. Please refer to individual product release notes for more information about supported features.

8.0.5 Client Control Vendor Response Requirement: Can the PC client be used to access and implement IPTS telephony features for control of the desktop telephone instrument?

Cisco Response: Cisco Unified Personal Communicator can be used to make calls using the integrated softphone or to control a Cisco Unified IP phone on Cisco Unified Communications Manager. Cisco recently announced Cisco UC Integration for Microsoft Office Communicator (MOC), a PC application that provides instant access to rich Cisco Unified Communications services (soft phone, mid-call control, messaging, conferencing, desk phone control, and phone presence) directly from a tab in Microsoft Office Communicator. (See Section 8.0.7 below for details).

8.0.5.1 Call Routing Control Call coverage and routing control should include, at minimum, the following programming parameters:

Caller priority level; Current station status Available route-to devices (desktop telephone instrument, mobile

teleworking client, wireless handset, cellular handset, et al) Time-of-day/day-of-week




Describe how the client GUI is used to screen, manage and route calls on a customized basis. Confirm that the above programming parameters are supported as part of the call routing control mechanism.

Cisco Response: Cisco Unified Personal Communicator provides powerful communications features integrated with your desktop or laptop 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:

• 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 send the call to voicemail with a simple mouse click.

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

• Presence: View real-time availability of other Cisco Unified Personal Communicator users. You can also choose to display customized messages, set out-of-office alerts, and show your availability based on your Microsoft Outlook calendar.

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

8.0.5.2 Voice (Speech) Portal Station uses should be able to use any type of telephone-type device to perform the following functions via voice commands:

Set presence status and preferred phone Access unified messaging system for voice and email messages Access Microsoft Outlook or IBM Lotus Sametime for contact lists,

calendar, tasks, and other personal folders Initiate or join scheduled and ad-hoc voice or web conference calls with

predefined workgroups Vendor Response Requirement: Describe voice portal features and functions available with the available UC solution, specifically addressing each of the listed voice command capabilities. Also identify other voice command operations supported by UC solution.

Cisco Response: IP Phone Messenger Network Interface: The IP Phone Messenger service included with Cisco Unified Presence 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. 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. 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.

8.0.5.3 Text-to-Speech The proposed UC solution should support a text-to-speech feature that converts text messages to speech format for access from various voice terminal devices such as desktop telephones and mobile handsets. Vendor Response Requirement: Confirm the proposed UC solution supports text-to-speech and briefly describe the communications operations it supports.

Cisco Response: Using the IP Phone Messenger Service users can 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. Unified Messaging: With the Text-to-Speech (TTS) capability of Cisco Unity Unified Messaging, you get information about all your messages -- and even hear the text portion of e-mail messages -- over the telephone.

8.0.6 Mobile Clients Vendor Response Requirement: What mobile clients are supported (Windows Mobile, Blackberry, et al) by the UC solution? What functions and operations can be supported on these mobile clients?

Cisco Response: Mobile workers using a Treo or BlackBerry device can simply double-click to play voice messages within their personal digital assistant (PDA) e-mail applications. The Cisco Unity solution supports a variety of notification options, including Short Message Service (SMS), e-mail, paging, and out-dialing, which allow you to customize the way you are notified of new voice messages. Cisco Unity Unified Messaging for Exchange users can access their voice messages using Cisco Unified Mobile Communicator, which integrates with Exchange to provide mobile access to messages. Even for users with basic mobile phones, the Cisco Unity solution is optimized to enhance mobile productivity. You can set up alternate phone numbers for mobile phones or other devices in the system to transfer calls to a mobile phone or speed login to the system. When calling in from a mobile phone, speech recognition allows for hands-free usage of the system. If a call is dropped because of a less than fully reliable mobile phone network, the Interrupted Session Recovery feature resumes, on the next call-in, the session where the call left off, reducing lost time. In addition, Cisco has included the Cisco Unified Mobile Communicator that is supported on Blackerry (830D, 8700g, 8703, 8800 and Pearl (8100)) and Nokia



(E61, E61i and E65) devices. Please refer to Section 4.6.2 “Advanced Mobile Client” above for additional details.

8.0.6.1 Mobile Security Vendor Response Requirement: What security features are supported for mobile clients?

Cisco Response: Secure messaging to enforce voicemail retention policies and prevent the compromise of voicemail messages with proprietary or confidential content forwarded to someone outside the enterprise.

• Password and PIN security policy options to enforce expiration, complexity, reuse, and lockout

• Optional RSA Secure-ID 2-factor one-time PIN authentication server interface

• Call-restriction tables to prevent toll fraud

• Security event logging and reports of failed login and account lockouts to help detect "PIN cracker" attack attempts

• SRTP and signaling encryption to ensure secure communication between the Cisco Unity system and Cisco Unified Communications Manager

8.0.6.2 Web Client Vendor Response Requirement: Is a Web-based client supported? If so, briefly describe its features and capabilities and how it differs from the standard UC client.

Cisco Response: The Cisco Unity Inbox is a browser-based message access console that provides a dedicated voicemail inbox that you can use to deliver unified-messaging functions to voicemail-only users. You can use the Cisco Unity Inbox interface to listen to any message in your voicemail inbox through either your PC or phone. With the Cisco Unity Inbox, you can receive notification of new voicemail messages right in your e-mail inbox using Simple Mail Transfer Protocol (SMTP) notification. Message notification provides an HTML link that you can click to automatically launch the Cisco Unity Inbox message access console, allowing you to play voice messages in WAV file format.

8.0.7 Microsoft Integration Vendor Response Requirement: What is the level of integration with Microsoft Outlook, Office and Active Directory? Is the proposed UC solution compatible with the scheduled Microsoft Office Communications Server (OCS) solution? Describe any hardware/software requirements and options required to integrate with the



Microsoft solutions.

Cisco Response: For more than ten years, Cisco and Microsoft have worked together to provide customers with innovative business solutions. Today, the two companies are partnering to help customers design Unified Communications solutions that closely align with their productivity, communication and collaboration objectives.

Cisco offers several categories of unified communications interoperability with Microsoft:

Cisco IP Telephony with Microsoft: If you want to use your Cisco Unified Communications Manager investments with Microsoft technologies, Cisco offers click-to-call capabilities from Microsoft Office Communicator (see below: “Cisco UC Integration for Microsoft Office Communicator (MOC)”), Microsoft desktop applications, and Windows Mobile handsets. With native simultaneous ringing capabilities of Cisco Unified Communications Manager, you can answer incoming calls on either Microsoft Office Communicator or your Cisco Unified IP Phone. In addition, you can enable Cisco Unified Communications Manager and Cisco Unified Communications Manager Business Edition to interoperate with Microsoft Exchange 2007 as a companion unified messaging solution.

Cisco Presence with Microsoft: Cisco also offers interoperability through

Cisco Unified Presence and Cisco Unified Communications Manager so you can view the presence status of Cisco Unified IP Phone users directly in the Microsoft Office Communicator client. Business-to-business federation between Cisco Unified Presence and Microsoft Live Communications or Office Communications Server facilitates ongoing collaboration between businesses. With Cisco Unified Personal Communicator, you can add Microsoft Office Communicator users in other businesses to your buddy list, view their presence status, and securely exchange instant messages for ongoing collaboration.

Cisco Messaging with Microsoft: For unified or integrated messaging with

Microsoft Exchange email, Cisco offers three proven enterprise-class solutions. Cisco Unity facilitates unified messaging with Microsoft Exchange so you can manage all email, voice, and fax messages in a single inbox. Cisco Unity Connection is an integrated messaging solution that offers access to voice messages from a Microsoft Outlook folder but requires less time and effort to deploy. And Cisco Unity Express provides integrated messaging with Microsoft Outlook for up to 250 users - ideal for small enterprises or branch offices.

Cisco Conferencing and Collaboration with Microsoft: For conferencing

and collaboration with Microsoft, Cisco offers the ability to schedule Cisco Unified MeetingPlace, Cisco WebEx, and Cisco TelePresence meetings directly from Microsoft Outlook. For real-time collaboration, you can initiate impromptu Cisco Unified MeetingPlace and Cisco WebEx meetings directly from Microsoft Office Communicator and use its presence and instant messaging capabilities. In addition, Cisco WebEx Connect also facilitates asynchronous collaboration by bringing together data from different sources (such as Microsoft Outlook and SharePoint) into one place.



Cisco Contact Center with Microsoft: The Cisco Unified CallConnector for Microsoft Dynamics CRM integrates Cisco Unified Communications with the Microsoft Dynamics CRM application at the desktop, without the need for additional hardware. It provides greater employee productivity and organizational efficiencies, immediate information about inbound and outbound calls, fast and easy click-to-dial functions from customer-relationship-management (CRM) database records, and call-duration tracking, information capture, and record creation.

Cisco Network with Microsoft: Cisco integrated services routers (ISR) and

Cisco Catalyst switches provide the ideal foundation for delivering unified communications applications for secure business-class, real-time communications and collaboration. Cisco integrated services routers are Microsoft-certified gateways for Microsoft Office Communications Server 2007 deployments.

With Cisco Unified Communications System Release 7.0, Cisco continues to execute on Cisco and Microsoft Interoperability roadmap by introducing:

Click-to-call from Microsoft desktop applications with Click to Call application – a Cisco Unified Communications Widget for your PC

Click-to-call from Windows Mobile smart phones with Cisco Unified Mobile Communicator 7.0

Ability to answer incoming calls on Cisco Unified IP Phone, Microsoft Office Communicator (MOC), and any other phone with simultaneous ringing capabilities of Unified Mobility natively available on Cisco Unified Communications Manager 7.0

Ability for Cisco Unified Personal Communicator 7.0 users in one business to exchange presence information and instant messages with Microsoft Office Communicator (MOC) users in another business enabled by business-to-business federation between Cisco Unified Presence 7.0 and Live Communications Server or Office Communications Server.

Ability to access Microsoft Outlook and SharePoint from Cisco WebEx Connect

Cisco UC Integration for Microsoft Office Communicator (MOC)

Cisco recently announced Cisco UC Integration for Microsoft Office Communicator (MOC), a PC application that provides instant access to rich Cisco Unified Communications services (soft phone, mid-call control, messaging, conferencing, desk phone control, and phone presence) directly from a tab in Microsoft Office Communicator.

Cisco UC Integration for MOC provides the following benefits:

Extends Cisco Unified Communications to Microsoft Office Communicator with easier-to-manage single call control architecture: Augments Microsoft Office Communicator’s IM capabilities with proven & reliable call control attributes of Cisco Unified Communications Manager—Quality of Service (QoS), Call Admission Control (CAC), and standards-based codecs support—without the cost



and complexity of simultaneously implementing and managing dual call control architecture.

Enables seamless (native to MOC) collaboration with Cisco Unified Communications and Microsoft instant messaging (IM): Delivers enterprise-class communications experience with Cisco IP phone like audio quality and calling capabilities, instant “VPN less” secure voice access, and Microsoft IM in the office, on the go, and from anywhere you can access your corporate network.

Provides investment protection and business-class communications with Microsoft: Facilitate business collaboration with industry leading Cisco Unified Communications while reducing your total cost of ownership (TCO) and protecting your investments in Microsoft presence & IM.

NOTE: numerous Cisco products also integrate with Active Directory, run on Windows and provide Web interfaces that can be accessed using Internet Explorer.

8.0.8 IBM Integration Vendor Response Requirement: What is the level of integration with IBM Lotus Sametime, Lotus Quickr, and Lotus Web conferencing? Describe any hardware/software requirements and options required to integrate with the IBM solutions.

Cisco Response: Together, Cisco and IBM are enabling a new way of communicating and collaborating – one that’s open, timely, and effective. Using best in class unified communications capabilities from both companies, you can transform your business processes and reach new levels of productivity.

Cisco offers several categories of unified communications interoperability with IBM:

Solution Overview Together, Cisco and IBM are enabling a new way of communicating and collaborating —one that’s open, timely, and effective. Using best in class unified communications capabilities from both companies, you can transform your business processes and reach new levels of productivity. Cisco Unified Communications with IBM Lotus Sametime provides deep integration among key components of the Cisco Unified Communications system and IBM Lotus Sametime. Lotus Sametime users can easily place voice or video calls and initiate integrated voice, video and Web collaboration sessions from their contact list and Instant Messaging (IM) sessions.

You can also see when someone is on the phone and access and manage voicemail directly from the Lotus Sametime client. These capabilities help save time, increase productivity and speed decision making by streamlining communications and enhancing collaboration.



Implemented as a suite of modular plugins, Cisco Unified Communications with IBM Lotus Sametime provides customers with the flexibility to use the full suite or choose the individual capabilities that best fit their needs. Cisco IP Telephony with IBM Lotus Sametime—For customers that want to initiate voice and video calls from Lotus Sametime, Cisco has integrated click-to-call capabilities. Cisco enables voice and video click-to-call capabilities by integrating Cisco Unified Communications Manager, Cisco Unified IP Phones and Cisco clients with Lotus Sametime. One option integrates the Lotus Sametime client directly with Cisco IP Communicator and Cisco Unified Video Advantage for voice and video calls that can be initiated from the contacts list and IM sessions. When this capability is used with Cisco’s voicemail integration with Lotus Sametime (see Cisco Unified Messaging with IBM Lotus below), you can easily return calls by clicking to call from voicemail records displayed in Lotus Sametime. Also available with this option is a phone control mode which lets Cisco Unified IP Phone users answer and manage incoming calls and initiate calls and conferences from Lotus Sametime. A separate click-to-call option is available for customers preferring a server-based integration with Cisco Unified Communications Manager that can be accessed from the basic call capabilities built into Lotus Sametime.

Cisco Presence with IBM LotusSametime—In addition to the presence status options provided in Lotus Sametime, users can also see phone presence indicators that show when a contact is on the phone. This capability lets you choose the most efficient way to contact a person you need to reach and avoid needless interruptions and playing “phone tag.” Cisco Unified Messaging with IBM Lotus Sametime—Through an integration with Cisco Unity® and Cisco Unity Connection, you can manage voicemail directly from the Lotus Sametime client. Capabilities include viewing a list of voicemails, playback and control of voicemails and filtering voicemails based on heard, unheard and deleted status. This integration helps users be more responsive by making them immediately aware of new messages.

Cisco Unified Communications with IBM Lotus Sametime



Cisco Unified Communication with IBM Lotus Sametime and IBM Lotus Notes

Cisco Conferencing with IBM Lotus Sametime—Through an integration with Cisco Unified MeetingPlace, customers can add conference setup to the presence and text messaging capabilities of Lotus Sametime. This lets you easily initiate and join voice, video and Web collaboration sessions directly from the contacts list or an IM session. The Web collaboration component can be configured to use either Unified MeetingPlace or WebEx. Lotus Sametime can be integrated with Lotus Notes such that Lotus Sametime shows up in a pane within the Lotus Notes client. The native capabilities of Lotus Sametime and the Cisco Unified Communications capabilities exposed through the plug-ins described above can be accessed from within Lotus Notes when it is deployed in this manner (see figure above). To get started customers can simply download the Cisco Unified Communications with IBM Lotus Sametime suite of plug-ins from Cisco.com and start integrating their Cisco and IBM Lotus user experiences today.

System Requirements The following table lists the Cisco Unified Communications with IBM Lotus Sametime system requirements broken out by the IBM Lotus components and each Cisco Unified Communications feature.

Component Requirement IBM Lotus Sametime Lotus Sametime Version 7.5.1 CF1 or later AND/OR IBM Lotus Notes Lotus Notes 8.0.1 or later IP Telephony - Cisco IP Communicator Click to Call with IBM Lotus Sametime*



Client Platform Windows XP (Service Pack 2), Windows Vista (Business, Enterprise; MSDN version is not supported)

Softphone Cisco IP Communicator 2.1(3) or later Cisco Unified Video Advantage 2.0 or later (optional)

Call Control Cisco Unified Communications Manager 4.1(3) or later Plug-in (client-side) Cisco Phone Control and Presence with IBM Lotus Sametime ** IP Telephony - Cisco Phone Control with IBM Lotus Sametime* Client Platform Windows XP (Service Pack 2), Windows Vista (Business, Enterprise;

MSDN version is not supported) Call Control Cisco Unified Communications Manager 4.1(3) or later (with CTI

enabled) Plug-in (client-side) Cisco Phone Control and Presence with IBM Lotus Sametime ** IP Telephony – Click to Call with IBM Lotus Sametime*v Client Platform IBM Lotus Sametime client platforms supporting TCSPI-based telephony

features Call Control Cisco Unified Communications Manager 6.0 or later Plug-in (server-side) Click to Call with IBM Lotus Sametime IP Telephony - Cisco Phone Presence with IBM Lotus Sametime* Client Platform Windows XP (Service Pack 2), Windows Vista (Business, Enterprise;

MSDN version is not supported) Presence Server Cisco Unified Presence 6.0(2) or later *** Call Control Cisco Unified Communications Manager 5.1 or later Plug-in (client-side) Cisco Phone Control and Presence with IBM Lotus Sametime ** Conferencing – Cisco Unified MeetingPlace Click to Conference with IBM Lotus Sametime Instant Messaging

Client Platform Windows XP (Service Pack 2), Windows Vista (Business, Enterprise; MSDN version is not supported)

Conferencing Server Cisco Unified MeetingPlace 6.0 or later Plug-in (client-side) Cisco Unified MeetingPlace Click to Conference with IBM Lotus Sametime

Instant Messaging Unified Messaging – Cisco Unified Messaging with IBM Lotus Sametime Client platform Windows XP (Service Pack 2), Windows Vista (Business, Enterprise;

MSDN version is not supported) Apple Mac OS X 10.4.x (PowerPC requires Sametime patch from IBM) Novell SUSE Linux Enterprise Desktop version 10 Novell Linux Desktop version 9 Red Hat Enterprise Linux version 4

Voicemail system (IMAP required)

Cisco Unity 4.2 with Microsoft Exchange 2003 Cisco Unity 4.2 with IBM Lotus Domino Cisco Unity 5.0 with Microsoft Exchange 2003 Cisco Unity 5.0 with Microsoft Exchange 2007 Cisco Unity 5.0 with IBM Lotus Domino Cisco Unity Connection 2.x

Plug-in (client-side) Cisco Unified Messaging with IBM Lotus Sametime

* “Cisco IP Communicator Click to Call with IBM Lotus Sametime” and “Cisco Phone

Control with IBM Lotus Sametime” are designed to be deployed together or individually with the other features in Table 1 except for “Click to Call with IBM Lotus Sametime.” Similarly, “Click to Call with IBM Lotus Sametime” is designed to be deployed with the other features in Table 1 except for “Cisco IP Communicator Click to Call with IBM Lotus Sametime” and “Cisco Phone Control with IBM Lotus Sametime.”

** Multiple features are delivered through this plug-in, and the administrator can

choose to enable any combination of them. Support for these features in Notes 8.0.1 is planned to be available in Q3 of calendar year 2008.



*** Cisco Unified Presence 6.x uses two Device License Units per user. Later versions require one Device License Unit per user.

8.1 Conferencing & Collaboration 8.1.1 Ad Hoc Conference Vendor Response Requirement: Is ad hoc and meet me audio/video conferencing supported by the available UC solution? Briefly describe how such a conference is established and managed.

Cisco Response: Cisco Unified MeetingPlace conferencing supports multiple interfaces for initiating impromptu or scheduling future rich-media conferences. In a single step, meeting organizers can schedule or initiate immediate voice, video, and Web resources through a Web interface, touch-tone and Cisco Unified IP phone, and Microsoft Outlook or IBM Lotus Notes calendar. Meeting invitees automatically receive notification by e-mail or calendar invitation and can attend rich-media conferences with a single click. Rescheduling meetings automatically reschedules all the resources at the same time. This simple approach saves time and improves productivity.

8.1.2 Scheduled Conference Vendor Response Requirement:



Can an individual station user schedule an audio/video conferencing session and reserve conferencing services as needed? Briefly describe how such a conference is established and managed.

Cisco Response: Users can schedule or initiate immediate voice, video, and Web conferences through a Web interface, touch-tone or Cisco Unified IP phone, Microsoft Outlook or IBM Lotus Notes calendar, and IM.

• Reservationless meetings: Users can set up impromptu voice, video, and Web conferences using a personalized meeting ID that is available to designated users. Reservationless meetings can be configured as internal or external meetings, and a maximum number of ports can be set for a reservationless meeting.

• Video scheduling and attendance: Users can schedule integrated rich-media meetings with voice, Web, and multiple video Cisco Unified Videoconferencing Multipoint Control Units (MCUs) and video terminals. They can search the directory to check the availability of specific video terminals that, when selected, are automatically outdialed from a Cisco Unified Videoconferencing MCU when the meeting starts. Users can also set up video-only meetings that connect through a Unified Videoconferencing MCU as well as select from multiple, preconfigured service codes that define bandwidth, video layout, and access restrictions.

• Collaborative, presentation, and Web seminar meetings: Cisco Unified MeetingPlace conferencing allows users to set up collaborative meetings for small, peer-oriented meetings such as project and staff meetings, presentation meetings for more structured Web conferencing meetings such as group training sessions, and Web seminar meetings for large, controlled meetings such as external presentations. In all meeting types, the meeting organizer can change user permissions as needed.

• Invitations: Calendar or e-mail invitations are automatically distributed to invited participants with the information needed to attend the integrated rich-media conference.

• Point-and-click attendance: Users attend voice, video, and Web conferences directly from their calendar, e-mail notification, URL link, IM, or browser.

• Autocascade for large meetings: When the number of participants exceeds the maximum number for a particular Web server, the system automatically cascades to other Web conferencing servers in the cluster to support a large meeting.

• Integrated IM: With Cisco Unified MeetingPlace Jabber Integration, users can easily initiate integrated voice, video, and Web conferences from Jabber Messenger. Similarly, with Cisco Unified MeetingPlace for Microsoft Office Communicator, users can initiate conferences directly from Microsoft Office Communicator. And with Cisco Unified Personal Communicator, users can initiate Cisco Unified MeetingPlace Web conferences.



8.1.3 Host Control Features An audio conference host should access and implement, at minimum, the following features/functions:

View conference participants; Play roll call of participants. Mute all lines; Mute select lines; Disconnect select participants; Dial out to bring new participants into the call; Record conference; Lock conference; Initiate a sub-conference.

Vendor Response Requirement: Briefly describe available audio conference host control capabilities, and confirm support of each of the above listed features/function (indicating implementation by TUI or Web-based control)

Cisco Response: Cisco Unified MeetingPlace provides the following in-session meeting features via the TUI:

– Announced entry and departure

– Roll call

– Breakout sessions (up to 9 sub-conferences)

– Mute individual users

– Mute all participants

– Outdial capability to an individual or a team

- Allows automatic recording and playback of meeting sessions

- Lock meeting

– Screened entry

All of the features mentioned above are available from both the TUI and the Web-based interface.

8.1.4 Web Collaboration Vendor Response Requirement: What form of Web services collaboration is provided for conference calls? Briefly describe how collaboration services are established and managed, including a listing of host control features and functions as they differ from audio-only conference capabilities. Indicate if recording and playback of an integrated audio/Web conference is supported.

Cisco Response: Cisco Unified MeetingPlace conferencing provides industry-leading in-meeting controls within the Web conference. Without disrupting a



meeting, users know who is attending, how users are attending (voice, video, or Web), who is speaking, and who is sharing. To help ensure that meetings run smoothly, users with the appropriate permissions can control a broad range of meeting characteristics, including Web conferencing permissions, speaking permissions (such as mute and listen only), recording, and meeting security. Users can control their own meeting environment, including muting and unmuting their phones, modifying their video layouts, and moving into private discussions. When a user chooses to leave or is ejected from a meeting, all media types-voice, video, and Web-are disconnected simultaneously. Having access to these integrated meeting controls allows meeting moderators to minimize disruptions, monitor meeting progress, and make rich-media meetings as natural and effective as face-to-face meetings. Users can send public and private text messages within collaborative meetings (refer to the discussion of collaborative, presentation, and Web seminar meetings later in this document). In presentation and Web seminar meetings, presenters can chat with anyone, and audience members can chat only with presenters. Users can automatically record and play back synchronized Web and voice meeting content without the need for additional hardware or software at the desktop. Voice recordings can be exposed as WAV, MP3, or Windows Media files. Cisco also recently introduced WebEx Connect, a new software-as-a-service (SaaS) platform that integrates presence, instant messaging, Web meetings and team spaces with traditional and Web 2.0 business applications. Please refer to Scetion 8.1.5 immediately below for a detailed description of WebEx Connect.

8.1.5 Application Sharing Vendor Response Requirement: Is application sharing services supported by the available UC solution? If yes list the specific applications.

Cisco Response: WebEx Connect is a new software-as-a-service (SaaS) platform that integrates presence, instant messaging, Web meetings and team spaces with traditional and Web 2.0 business applications. It is the first collaboration-centric SaaS platform that allows developers and customers to integrate multiple applications to create powerful collaborative business “mash-



up” solutions from best in class web and desktop applications. Developers and partners can extend their reach by offering innovative collaborative solutions to millions of WebEx users worldwide. WebEx Connect integrates with WebEx meetings applications and the Cisco Unified Communications portfolio for transparent communication and collaboration.

Benefits of WebEx Connect:

WebEx Connect facilitates business collaboration across firewalls while enforcing enterprise-class security, scalability, performance, and availability.

WebEx Connect provides an open and extensible collaboration platform that allows for easy integration, so you can continually enrich and extend the value of your applications and business processes.

WebEx Connect offers easy customization, deployment, and administration with low maintenance and an on-demand delivery model.

WebEx Connect works with the Cisco Unified Communication Solution to deliver transparent communication and collaboration.

Detailed information is available at: http://www.cisco.com/web/about/ciscoitatwork/trends/webex_connect_workforce_exp/index.html Cisco MeetingPlace: Application and desktop sharing: Users can share any document or application or their desktops (including dual monitor systems) from Windows (Internet Explorer, Firefox, or Netscape browsers) and Mac OS (Safari or Firefox browsers). Shared content automatically resizes to the viewer's desktop resolution. Users can share multiple applications at the same time within the meeting for side-by-side viewing, as well as participate in multiple Web conferencing sessions at the same time. Presentations: Users can upload and share PowerPoint presentations, graphics files (jpg files), Flash content (swf files), or Flash movies (flv files) using a Web browser under Windows and Mac OS. Animations within the presentations are preserved as slides are advanced. Annotations and whiteboards: Users can annotate shared applications, presentations, and whiteboards. Multiple whiteboard sessions are available per meeting.

8.2 UC Architecture Vendor Response Requirement: Briefly describe the hardware/software architecture of the available UC solution, including all necessary server and client software requirements. Address in your response the following:

Embedded open industry standards ands specifications, such as SIP,



SOA Web services, et al. Level of design redundancy for both hardware and software elements,

network interfaces, and memory storage Subscriber system capacity and scalability parameters Capability to support multiple PBXs (IPTS) and remote sites

Cisco Response: Cisco Unified Communications Solutions unify voice, video, data, and mobile applications on fixed and mobile networks, delivering an easy-to-use, media-rich collaboration experience across business, government agency, and institutional workspaces. These applications use the network as the platform to enhance comparative advantage by accelerating decision time and reducing transaction time. The security, resilience, and scalability of the network enable users in any workspace to connect, everywhere, every time, everyone’s connected. Cisco Unified Communications is part of a comprehensive solution that includes network infrastructure, security, wireless, management, lifecycle services, flexible deployment and outsourced management options, and third-party applications. All of the components quoted to meet the unified communications requirements are described in detail in Section 1, including the servers, clients, and redundancy provided. The unified communication system scales to well over the 3,000 user growth requirement of VoiceCon.

8.3 Systems Management UC systems management functions should include, at minimum, the following:

Station user telephone instrument management Status displays for all UC system components Audit trails (configuration log) System resource monitoring (CPU, memory, disk space) Serviceability, error logging, and tracing features Storage & Reporting features Interface for backup and recovery

Vendor Response Requirement: Briefly describe UC systems management tools, including hardware/software requirements, and confirm support for each of the above operations. Identify operations not supported

Cisco Response: The management tools provided for these unified communications applications are described in Section 6.

8.4 Security What security features and functions are embedded or are available with the available UC solution? In your response specifically address each of the following security issues:

Password Access Authorization



Authentication Encryption Certification strategy

Cisco Response: Cisco’s Unified Communications security strategy is identical to that described in section 1.6 above. In addition, the Cisco Unified Personal Communicator uses username and passwords to authenticate to the Cisco Unified Presence server and digest authentication to authenticate to Cisco Unified Communications Manager. Cisco Unified Presence allows users to configure rules that control who is able to view their presence and exactly how their presence is represented based on a number of factors. Cisco Unified Personal Communicator provides easy access to Cisco Unified MeetingPlace, Cisco Unified MeetingPlace Express, Cisco Unity and Cisco Unity Express. All of these unified communications applications require credentials from that user to authenticate them before services can be rendered. All configuration- and preferences-related communications between Cisco Unified Personal Communicator and Cisco Unified Presence is done over TLS to provide per packet authentication, integrity and confidentiality.

.



Section 9: Automatic Call Distribution (ACD) Contact Call Center

9.0 Basic ACD Call Center Requirements

VoiceCon requires that the proposed IPTS communications solution must support an ACD-based contact call center solution that includes call screening, call prompts, automatic distribution routing, call queuing, announcements, call handling, agent mobility, management and reporting, feature configuration and programming, administration.


Confirm your proposed communications system can support the listed functional requirements and briefly describe the necessary hardware and software requirements. Indicate if any ACD capabilities are embedded in the IPTS generic software package.

Cisco Response: Cisco Unified Customer Contact solutions provide powerful collaboration tools that transform customer care from simple phone transactions to unique, rich experiences that can be personalized for individual customers.

Build competitive advantage: Contact center agents can use a variety of rich media including voice, web, e-mail, and video to provide personalized, customer-centric services.

Accelerate time-to-resolution: Advanced communications capabilities maximize agent and contact center productivity.

Enhance customer satisfaction: Powerful, self-service solutions help to enhance the overall customer experience while decreasing contact center costs.

Increase revenues: The ability to use advanced collaboration tools and rich media lead to more up-sell and cross-sell opportunities.

Improve first call resolution: No matter where they're located, presence-based knowledge workers in your business can assist and collaborate with your customers.

Detailed information about Cisco’s Contact Center product is located at: http://www.cisco.com/en/US/products/sw/custcosw/Products_Sub_Category_Home.html New and Enhanced Products are described in detail at the following links:

Cisco Unified Contact Center Enterprise and Hosted 7.5 Cisco Unified Contact Center Express 7.0



Cisco Unified Customer Voice Portal 7.0 Cisco Unified Call Studio 6.0 Cisco Unified Call Services, Universal Edition 6.0 Cisco Unified Expert Advisor Cisco Unified Intelligence Suite 7.5 Cisco Agent Desktop 7.0 and 7.5

To meet the requirements of this RFP, Cisco would propose the Cisco Unified Contact Center Enterprise. The Cisco Unified Contact Center Enterprise provides a complete, multi-channel skills-based routing contact center as part of Cisco’s Unified Communications solution. Designed to meet the challenging business needs of today’s contact centers, Contact Center Enterprise supports advanced call treatment, queuing, customized announcements, and agent desktop CTI to effectively mange customer contacts across all media types. Agent mobility is included with options from simply extending the system over IP to remote sites, to at-home workers using Cisco’s proven Enterprise Class Teleworker solution or even agents working remotely using their own local POTS lines to handle contacts. Complete management and both real-time and historical reporting is also included as part of the system.

The Contact Center Enterprise works with Cisco’s powerful Unified Communications Manager to provide a robust and highly scalable telephony platform for the contact center agents. Queuing is provided in the Cisco Unified IP-IVR or Customer Voice Portal platforms. Core routing of contacts is managed by the Intelligent Contact Management (ICM) with proven integration to leading 3rd party products like wall boards, workforce management, quality management and other key contact center applications.

9.01 Third Party System Integration

VoiceCon requires that the proposed ACD solution be able to support third party equipment.

Vendor Response Requirement: Confirm that the proposed ACD solution can support each of the following third party equipment options:

Message Board

Interactive Voice Response (IVR) system

Customer Relationship Management (CRM) system

Workforce Management system

Quality Monitoring system

Cisco Response: Cisco Unified Contact Center Enterprise has proven integrations with the leading 3rd party applications for contact centers. Cisco provides open and standards-based interfaces and APIs to allow these partners to access key contact center components.



Message Boards. Cisco’s partners leverage the open SQL database to gather and display real-time performance metrics for the contact center – using ODBC to access the data from their core application platform.

Interactive Voice Response. Cisco’s IVR partners have a standard interface (known as GED-125) to provide message-level integration with the contact center as well as allow callers to opt-out of the IVR and be routed intelligent, with full call context, to agents in the contact center.

Customer Relationship Management. Cisco’s CRM partners offer integration to Cisco’s contact center using the open CTI interface (GED-188) to agents to work in the CRM system’s native web or client application and perform call and agent state control functions without ever touching their phone. Additionally, Cisco offers pre-integrated adaptors for leading CRM vendors like: Siebel, Oracle, PeopleSoft, SAP, Salesforce.com and others.

Workforce Management System. Cisco’s WFM partners offer integration to Cisco’s contact center for both historical data retrieval from the system as well as real-time adherence tracking with the open CTI interface to monitor agent adherence to schedule in real-time.

Quality Monitoring System. Cisco’s QM partners offer integration to Cisco’s contact center to be able to match call context data to their recordings in real time using the open CTI interface.

9.0.2 ACD Station Equipment

VOICECON requires that the proposed ACD solution be capable of supporting a mix of terminal equipment for agents and supervisors.

Vendor Response Requirement: Confirm that the proposed ACD solution can support a mix of ACD agent/supervisor station equipment that includes analog and IP desktop telephone instruments and PC client soft phones for agent/supervisor voice communications requirements.

Cisco Response: Cisco does not manufacture a specific ACD or “call center” phone, the contact center agents and supervisors can leverage the organization’s investment in the standard Cisco IP Phones that all users on the system would access. The specific Cisco IP Phone models supported in Contact Center Enterprise include:

Cisco Unified IP Phone 7970/7971/7975

Cisco Unified IP Phone 7960/7961/7962/7965



Cisco Unified IP Phone 7912


IP Communicator (desktop soft phone client)



These phones can use any mix of supported protocols, including SCCP, SIP, or H.323 – and a contact center can use any mix of the supported phones or protocols across the system.

9.0.2.1 ACD Telephone Instrument

Vendor Response Requirement: Briefly identify and describe all desktop telephone instruments and equipment in your portfolio that are designed specifically for ACD agents and supervisors.

Cisco Response: Cisco does not manufacture a specific ACD or “call center” phone, the contact center agents and supervisors can leverage the organization’s investment in the standard Cisco IP Phones that all users on the system would access. The specific Cisco IP Phone models supported in Contact Center Enterprise include:







IP Communicator (desktop soft phone client)

Supervisor Workstation

Vendor Response Requirement: Briefly describe the proposed supervisor workstation solution in your ACD solution proposal, including telephony and ACD-specific feature and functions, toolbars, and report screens.

Cisco Response: The Cisco Contact Center Enterprise provides for both Agent and Supervisor desktop applications – using either our “out of the box” Cisco Agent Desktop (CAD) or toolkit-based CTI OS developer kit.

The Supervisor desktop allows contact center supervisors and managers to monitor the activity across their teams – with specific views into the calls being processed by agents, the agent states and the queues associated with their teams. Supervisors can use the desktop to proactively manage their agents by changing their agent state - for example, making an agent available that has been in “not ready” state too long. Supervisors can also use the desktop to silent monitor their agents, listening in on active calls and provide coaching via text chat to the agent directly to their desktop client. The supervisor can also barge into active calls, as well as intercept a call to take a call over from the agent – perhaps to provide additional services to the caller.

The Supervisor desktop also provides real-time reports of contact center activity, with built-in threads holds to show critical values in different colors, making it easier for the supervisor to quickly spot changes in volume or call



handling and take action before things get out of hand. The desktop can also provide “Supervisor Workflows” to not only display information, but send alerts to the supervisor when they are away from their desktop – enabling more active response faster based on the contact center’s business rules.

In this screen shot, the Cisco Agent Desktop (CAD) Supervisor Desktop shows the skill groups associated with the supervisor’s teams – as well as the agents and their states/active calls. In this example, the agent Pete Wall’s inbound call is selected, allowing the supervisor to see the same information the agent sees on their desktop for the real-time call display, enterprise data (screen pop data) and call history.



In this sample screen, the Supervisor Desktop is showing three “tabular” and one “graphical” report – driven from real-time data of the agent team for the Supervisor.

9.0.3 Remote ACD Station Equipment

VoiceCon requires that the proposed ACD solution can support remote agents.

Vendor Response Requirement: Confirm that the proposed ACD solution can support remote off-premises IP desktop telephone instruments and/or PC client soft phones for agent communications requirements. Also indicate if your system solution can guarantee Quality of Service levels for PSTN level connections to remote agents.

Cisco Response: The Cisco Contact Center Enterprise supports agents across multiple locations, treating them as part of the virtual contact center no matter where or how they are physically connected. Agents can be remote at on-network facilities or buildings on a campus using their Cisco IP Phone or desktop IP Communicator client on their laptop or PC. At home agents can be supported as well using the Cisco Enterprise Class Teleworker solution to extend a secure VPN connection to a remote user over a dedicated WAN or shared/Internet connection. In this model, agents would use their Cisco IP Phone at home behind a Cisco 7xx-series VPN Router, which manages the secure, QoS-enabled network connectivity across the network. Additionally, the system can support agents who do not have a high speed data connection that is “voice ready,” the system can leverage their existing home phone service (POTS), re-directing their contact center calls to their phone or any designated phone number.

9.0.3.1 Virtual Contact Center

VoiceCon may require a virtual contact call center solution in the future as it integrates CHQ facilities with other corporate facilities.

Vendor Response Requirement: Confirm that your proposed ACD solution can support a virtual call center environment, and include in your response answers to the following questions:

How many remote sites can be supported?

How remote sites are physically and logically supported?

Can call loads be balanced across multiple sites to avoid agents sitting idle at one site while other sites are overloaded?

Are all call center operations transparent across sites for ACD call routing, supervisory and reporting functions, telephony features, and any applications such as call recording?

Can your system guarantee Quality of Service (QoS) at the PSTN level for the voice channel for remote agents, and if so how is this done?



Are remote agents measured, service observed, recorded the same as local agents? Can they be members of the same ACD group, queue, split/skill as local agents?

Cisco Response: Cisco’s Contact Center Enterprise solution is designed to be highly scalable and flexible, supporting a number of different deployment models to meet the demands of the organization. To the specific question points above:

How many remote sites can be supported?

Cisco Response: There is no limit to the number of remote sites that can be supported by the Contact Center Enterprise system. They can be remote sites where the phones are “homed” back to a Unified Communications Manager cluster at a central data center, or the site may have its own local Unified Communications Manager – the agents are all considered part of the same virtual contact center across all the sites and clusters.

How remote sites are physically and logically supported?

Cisco Response: Physically, remote sites are simply sites connected via an appropriate IP WAN connection. The remote site may support only a few agents or thousands of agents. Logically, the sites can be mapped by creating their own “site-based” skill groups if the organization wants to track and report based on individual sites – these site-based groups can also be automatically rolled up into an “Enterprise” group for routing and reporting that makes multiple sites look like a single site, while still having the detailed data to “drill down” as needed.

Can call loads be balanced across multiple sites to avoid agents sitting idle at one site while other sites are overloaded?

Cisco Response: The Contact Center Enterprise is a virtual contact center – regardless of the number of sites, contacts are queued under control of a single routing engine – which monitors the state of all agents across all sites. Calls can be routed to any agent at any site at any time by the system—based on the business rules of the organization. Using this virtual network queue, there is no concern about sites being overloaded or calls getting “stuck” in a queue at one site when agents become available at another site.

Are all call center operations transparent across sites for ACD call routing, supervisory and reporting functions, telephony features, and any applications such as call recording?

Cisco Response: Contact Center Enterprise allows for the central administration and control of the system as a single virtual application, across all sites and functions. Supervisors can monitor their agent teams no matter where those agents physically are located.

Can your system guarantee Quality of Service (QoS) at the PSTN level for the voice channel for remote agents, and if so how is this done?

Cisco Response: Contact Center Enterprise leverages the power of the Cisco Network to ensure voice quality using standard Call Admission Control



functions, as well as enabling QoS tagging at the application layer for all of the system components.

Are remote agents measured, service observed, recorded the same as local agents? Can they be members of the same ACD group, queue, split/skill as local agents?

Cisco Response: Contact Center Enterprise provides the same administrative and supervisor functions for agents no matter where they are located – teams can include agents who are local in the “brick-and-mortar” call center, at remote sites on a campus or across the corporate network, as well as at home. There is no restriction on the agent location to enable flexible/mobile workers in the contact center.

9.04 Redundancy

VoiceCon requires a high level of service availability for its contact center solution.

Vendor Response Requirement: What levels of redundancy are embedded into your ACD solution design? Be specific as to ACD call control and routing functions, switched connections, announcements, and MIS reporting capabilities. Identify if redundancy is based on fully duplicated or load sharing hardware or software elements.

Cisco Response: Cisco Contact Center Enterprise is designed to provide “carrier class” fault tolerance with no single point of failure. The system leverages the clustering redundancy of Unified Communications Manager – in single sites as well as using Clustering over the WAN. Caller treatment and queuing functions also are supported in redundant configurations, automatically routing around failed components and recovering without manual intervention. The core routing engine, ICM is deployed redundantly as well with the Call Router platforms deployed either co-located or geographically distributed to provide spatial redundancy as well.

9.1 ACD Contact Call Center Parameters

VoiceCon requires an ACD contact call center capability to support the following basic parameters:

115 active and 200 configurable agents with workstations

5 active and 10 configurable supervisors with workstations

15 agent groups/splits

50 integrated announcements with scripts

Confirm the proposed communications solution can satisfy these parameter requirements, and complete the following table.

Proposed ACD System Parameters Maximum # Configurable Agents with workstations Unlimited Maximum # Active Agents with workstations 8,000

Concurrent



Maximum # Configurable Supervisors w/workstations

Unlimited

Maximum # Active Supervisors w/workstations Supervisors are included in the Agent Count

Maximum # Splits/Groups Unlimited Maximum # Trunk Groups Unlimited Maximum # ANI/DNIS numbers Unlimited Maximum # Call Delay Announcements Limited only

by Disk Space for storage of .WAV file

Maximum # Queue Slots Unlimited Standard MIS Reports 120+ # Real Time 40+ # Historical 60+ # Exception 20+

9.2 Basic ACD Features

The proposed ACD solution should include, at minimum, the following basic features:

Agent mobility (e.g. login at any terminal); Multiple Agent Groups Call Flow Applications/Skills-based Routing Priority Queuing Call Overflow and Interflow Redirect on No Answer Predictive Overflow Recorded Announcements Music between Recordings Dial out of Queue Work codes Work timers Make Agent Position Busy Agent Help Request to Supervisor Silent Monitoring – Split Monitoring: Supervisor & Agent Silent Monitoring – Split Monitoring: Supervisor, Agent and Caller Threshold Alerting Queue status



Vendor Response Requirement: Confirm that the proposed ACD solution supports each of the listed features/functions, identifying any feature/function not supported:

Cisco Response: Cisco Contact Center Enterprise supports all of these features/functions as follows:

Agent mobility (e.g. login at any terminal) Cisco Response: Cisco Contact Center Enterprise provides support for agent mobility both inside and outside the contact center. Within the contact center, agents can leverage Cisco’s Extension Mobility feature to log into any phone with their profile and have their extensions, speed dials and other functions follow them to the new phone. Agents can also “free seat” using the Cisco Agent Desktop client, which allows them to enter the extension they are using for their session, without having to have a fixed extension per agent. Outside the contact center, agents can use our Mobile Agent functionality to use their home phone, cell phone or any direct dial number as their agent line and still be part of the virtual contact center across the enterprise.

Multiple Agent Groups Cisco Response: Cisco Contact Center Enterprise allows for the configuration of multiple agent groups in the system and for individual agents to be assigned to multiple groups within the system.

Call Flow

Cisco Response: Cisco Contact Center Enterprise provides a graphical “Script Editor” application that allows the administrator to define specific call flow logic using a pallet of call flow objects, like time of day decisions, IF/Then logic, queuing, etc. based on real-time conditions in the contact center. The call flows can also be monitored from Script Editor in real-time, displaying the number of calls in the flow and the real-time stats for the related agent groups, call types and applications.

Applications/Skills-based Routing

Cisco Response: Cisco Contact Center Enterprise provides for advanced skills-based routing functionality in the core application, supporting functions like look-ahead queuing/routing, overflow/backup agent groups, and longest available agent across a set of skill groups.

Priority Queuing Cisco Response: Cisco Contact Center Enterprise provides for calls to be assigned a queue priority based on a numerical value (1 – 10) that can be assigned when a call is initially queued and then incremented or decremented over time while the call is in queue based on any real-time condition or customer status value. For example, all callers might be queued at a level 5 (five) but the script can check the status of the caller and automatically “bump up” the queue priority for “gold” callers in the call flow.



Call Overflow and Interflow Cisco Response: Cisco Contact Center Enterprise treats each call individually, allowing the call flow logic to automatically expand the agent / skill groups the call is queued for as the call approaches the point of breaking service level – overflowing the call into another group in the system, while maintaining the original position in queue for the initial selected groups. Interflow tends to be a concept for multiple sites where a call is queued across multiple sites at the same time; however in the Cisco solution, there is only one virtual network queue for all agents across all sites, eliminating the need to interflow calls.

Redirect on No Answer Cisco Response: Cisco Contact Center Enterprise automatically handles “redirect on no answer” calls where an agent was targeted for a call but the agent did not answer the contact. This is handled automatically within the Customer Voice Portal (CVP) application, selecting the next available agent for the caller without losing the position in queue. The agent that did not answer the call initially is automatically set to “not ready” and tracked with a reason code for the state change.

Predictive Overflow Cisco Response: Cisco Contact Center Enterprise call flow logic has full access to the real-time and historical conditions of the contact center for routing of calls – this allows the system to predict based on the past performance how long a caller might have to wait in their target queue to be answered and automatically overflow the call into additional agent / skill groups to ensure the contact is answered within the service level goal for that call type.

Recorded Announcements Cisco Response: Cisco Contact Center Enterprise uses standard .WAV files to store announcements in the system for callers, which are placed on standard HTTP web servers and called by the Customer Voice Portal (CVP) VXML applications during the call flows. Since each call is treated individually, the system can provide a wide range of announcements to different callers based on their status, the number of calls in queue ahead of them, time of day, etc. There is no limit to the number of announcements for the system.

Music between Recordings Cisco Response: Cisco Contact Center Enterprise allows the administrator to define the music sources and message/announcements played to callers in the system using standard .WAV files or with streaming media (RTSP) feeds on the network.

Dial out of Queue Cisco Response: Cisco Contact Center Enterprise supports “courtesy call back” to allow callers in queue to be offered the choice of staying in queue or having the system call them back either on a scheduled basis or when their call would normally have been returned. Additionally, while callers are in queue, they can opt to perform self-service functions (check balances, listen to FAQ’s, etc.) while retaining their position in queue waiting for an agent.



Work codes Cisco Response: Cisco Contact Center Enterprise captures multiple codes associated with the agent and caller – Not Ready Reason Codes for tracking agents making themselves not ready for calls in the system, Wrap-up Codes for tracking specific actions associated with calls, and Logout Reason Codes for tracking agents logging out of the system. The system allows these codes to be turned on/off at the agent profile level, allowing the flexibility of having specific groups use Wrap-up Codes and others not to have to enter them into the system. All codes are tracked in the system database for reporting purposes.

Work timers Cisco Response: the agent activities such as logged in time, not ready time, hold time, talk time, etc. These metrics are displayed in real-time as well on the Cisco Agent Desktop for the agent to be able to track the time spent on a call, as well as on the Supervisor Desktop as real-time statistics to help monitor the agent and call activity.

Make Agent Position Busy Cisco Response: Cisco Contact Center Enterprise allows the agent to control their “state” and go “not ready” from the Cisco Agent Desktop. Supervisors also can control the agent state from their Supervisor Desktop to force the agent ready/not ready.

Agent Help Request to Supervisor Cisco Response: Cisco Contact Center Enterprise provides two levels of “emergency assist” at the Cisco Agent Desktop – “supervisor” and “emergency.” The Supervisor assist button on the desktop automatically looks for a supervisor for the agent and joins them to the call automatically as well as opens a separate agent/supervisor chat window to allow the agent and supervisor to communicate “out of band” about the reason for the escalation to the supervisor. The Emergency assist button on the desktop automatically conferences in the supervisor and can send a CTI event to the call recording system to begin recording this interaction for lega/regulatory compliance. These functions can also be disabled in the system at the agent profile level.

Silent Monitoring – Split Monitoring: Supervisor & Agent Cisco Response: Cisco Contact Center Enterprise provides Silent Monitoring functionality from the Supervisor Desktop application, presenting the Supervisor with a list of active calls for all of their agents and allowing from a single button on the desktop to be silently joined to the call. From the monitoring session, the supervisor can barge into the call, joining as part of a conference or intercept the call away from the agent to allow the supervisor to take over the interaction.

Silent Monitoring – Split Monitoring: Supervisor, Agent and Caller Cisco Response: Cisco Contact Center Enterprise provides a “Remote Silent Monitor” functionality that allows remote supervisors or managers to dial into the system, be authenticated by the system, and listen to calls in real-time



over their phone-in connection.

Threshold Alerting Cisco Response: Cisco Contact Center Enterprise provides threshold altering in the standard reporting package, highlighting values that exceed (or are under) a given threshold on any real time or historical report in the system. Additionally, the Supervisor Desktop can take specific actions based on real-time conditions such as sending a message to all the agent desktops in the contact center, sending an e-mail or page to the supervisor or manager, etc. The real-time views on the Cisco Agent and Supervisor desktops can also provide visual indication of a stat exceeding (or being under) a given threshold automatically as well.

Queue status Cisco Response: Cisco Contact Center Enterprise provides real-time queue status, tracking the number of calls in queue, oldest call in queue, number of abandons in real time as well as on an user-defined interval basis to show queue stats over a range of time buckets.

9.3 Call Flow

At minimum the proposed ACD contact call center solution must be able to provide call control, screening, and routing based on:

Incoming Trunk Group ANI, DNIS, or CLID Call Volume System Performance Criteria Priority Queuing Call Prompts

Vendor Response Requirement: Confirm that the proposed ACD solution supports each of the listed call control criteria and briefly describe how supervisors/administrators create and develop scripts for incoming call flow operations.

Cisco Response: Cisco Contact Center Enterprise allows contact flows to be developed that take call information as well as real-time conditions into consideration when processing the calls. Caller prompting and databases can also be used to provide caller segmentation to determine the proper routing of the contacts in real-time based on the organizations business rules. Administrators can create and develop call flow routing scripts using the Script Editor tool. This tool can be accessed on the Admin Workstation of the system, or downloaded to the user’s laptop or Personal Computer to allow them to work anywhere they have a network connection to the system – at home, at a remote office, or even at a local coffee shop using a secure wireless VPN connection. The Script Editor tool is a graphical tool, which provides the user with a set of



routing tools in a pallet that they can drag and drop into the workspace. The tools become nodes in the script, and these notes instruct the system to perform actions – such as the “Run External Script” node tells the system to play an IVR script to the caller, a Queue to Skill Group node tells the system to look for an agent in one (or more) skill groups and monitor those agents for this call. The nodes are connected to create a logical flow – like a flow chart. Nodes have “success” and “failure” branches, so if a Queue to Skill Group node included a “Sales” skill group where no agents were logged in, the node would “fail” and pass control to the next node down the “failure” path. Script Editor also includes a tool to validate scripts, as only valid scripts are allowed to be saved and activated in the system. The system can also maintain a user-defined number of script revisions, allowing for an easy “roll-back” to a prior script version if necessary. Scripts can also be tested using the Call Tracer tool which simulates a call in the script, using the real time conditions of the contact center to help verify the logic of the script.

9.3.1 Routing & Queuing

Vendor Response Requirement: Confirm the proposed ACD solutions can support, at a minimum, each of the following call flow routing & queuing decision criteria, identifying any not supported.

First In/First Out (FIFO) Time of day (TOD) / Day of week (DOW) / Day of year (DOY) ANI/DNIS/CLID Originating call by voice terminal type, i.e., cell phone or payphone Call prompt response Number of calls in queue Abandoned Calls Longest held call in queue Estimated wait time Available agents (number, skill) Agent idle time Agent handle times Caller directed routing

Cisco Response: Cisco Contact Center Enterprise can provide routing of contacts based on all of these specific conditions with the exception of the “voice terminal type.” This type of information is dependant on the carrier network being able to provide the “II” or Informational digits about the call in the call signaling from the carrier network. If this information is provided by the carrier, the Cisco Customer Voice Portal can pre-screen these calls when they arrive at the Cisco Voice Gateway to provide unique treatment.



9.3.2 Agent Skills Vendor Response Requirement: Confirm that the proposed ACD solution supports skill-based routing, and identify:

Total number of programmable skills; Number of programmable and active skills per agent; Number of programmable and active multiple group assignments; If an agent can add/delete their skill assignments;

Cisco Response: Cisco Contact Center Enterprise supports skills-based routing with the following parameters:

o Total number of programmable skills; The number of skills that can be configured/programmed into the system is unlimited

o Number of programmable and active skills per agent; Agents are limited to 50 skills per agent in the 7.5 release of the Contact Center Enterprise system

o Number of programmable and active multiple group assignments; There is no limit to the number of “Enterprise” skill groups that can be configured/programmed in the system

o If an agent can add/delete their skill assignments;

Agents can not modify their own skills unless they are given access to the agent-reskilling tool included with the system. Typically, only supervisors use this tool to manage the skill groups associated with their agents.

9.3.3 Customer Preference

Vendor Response Requirement: Confirm the proposed ACD solution supports customer preference call routing and queuing capabilities, and briefly describe the process how incoming callers can control their call flow.

Cisco Response: Cisco Contact Center Enterprise supports customer preference call routing, allowing callers to pre-define their preferences in an external database to perhaps define on a per customer basis a specific agent or agent group that should be associated with the specific customer and/or organization. Callers can also be prompted to select a specific service – sales, service, etc. While in queue, callers an also be prompted to allow them to perform self-service functions while retaining their position in queue.



9.4 Call Processing Functions

Provide brief answers to each of the following questions regarding the proposed ACD solution call processing features and operations.

9.4.1 Describe how call control and agent handling methods can be administered locally and changed on demand if necessary in response to system activity.

Cisco Response: Cisco Unified Contact Center Enterprise can dynamically change the way contacts are handled in the routing scripts based on real-time conditions. Contacts are handled on a call-by-call basis, which means the first five calls into a queue might get the same treatment and messages but the sixth call may get a different set of messages due to the queue length or expected wait time, without impacting the calls already in queue. Administrators can also set these sorts of thresholds or decision points using global variables which they can modify in the scripts manually or even set up IVR applications to allow them to dial in remotely and change the variables, perform “emergency” closes or take other actions at any time.

9.4.2 How many priority levels can be assigned incoming calls? Can priority level be changed while it is in queue based on system factors, i.e., time in queue, available agents, etc.?

Cisco Response: Cisco Unified Contact Center Enterprise supports 10 call priority levels (1 – 10) Calls are assigned a priority at the time they are queued, in the routing script. The priority level can be changed at any time in the script as well based on system load, time in queue, or caller segementation.

9.4.3 How many callers can concurrently listen to a particular ACD recorded announcement? Is the number based on origin of the announcement, i.e., internal or external to ACD system?

Cisco Response: There is no limit on the number of simultaneous callers listening to an announcement as the announcements are stored as .WAV files and cached in the system automatically to play to callers individually in real-time.

9.4.4 Can announcements played for a caller be defined as “uninterruptible” even when agents are available to handle the incoming call?

Cisco Response: Yes, announcements can be defined as “uninterruptible” for legal and service observe warnings that all callers must hear before being connected to an available agent.

9.4.5 Describe all available and standard automated prompt features that a caller would have to listen and react to for call screening and routing procedures.



Cisco Response: The call flow scripts in the system define which prompts and announcements are played to callers in real-time. Call screening could be done based on looking up the caller’s phone number in a contact database—without having to prompt for an account number or other contact details to provide routing for that caller. This can be user defined to meet the specific business needs.

9.4.6 Can callers maintain their position in queue while interacting with an IVR?

Cisco Response: Yes, the call flow script can allow callers to interact with self-service flows while maintaining their position in queue for an agent.

9.4.7 Describe any other unique call routing features available on your ACD system that you believe VoiceCon would be interested in knowing about.

Cisco Response: Cisco Contact Center Enterprise allows for the creation of contact routing scripts that can address the most complex business requirements – using real-time conditions in the system to predict wait times and automatically overflow callers into backup groups as well as considering agent occupancy and workload when selecting agents to service callers. Other ACD vendors sell advanced routing packages to provide similar routing functionality; however, with Cisco this is included in the core application at no additional charge.

9.5 Supervisor Functions

The proposed ACD solution must be able to support supervisor positions to monitor and assist call agent positions, monitor and review system performance, and administer ACD functions and operations.

Provide answers to each of the following supervisor function requirements.

9.5.1 Describe how a supervisor can remotely monitor an agent. Indicate if the agent is notified by the system if they are being monitored by a supervisor, and specify the type of notification signal.

Cisco Response: Cisco Contact Center Enterprise allows supervisors to remotely silent monitor their agents using their desktop application or by dialing into the system from any location. Agents are not made aware of the observation, but supervisors can use the built-in chat/coaching function to let the agent know they are being monitored if necessary.

9.5.2 Can a supervisor assist an agent during an active call, and if so can the agent received prior notification this is about to occur?

Cisco Response: Cisco Contact Center Enterprise allows supervisors to assist agents with the “barge in” function from their Supervisor Desktop application. The agent will see the supervisor joining the call as a “forced conference” both on their phone and in their agent desktop window.



9.5.3 Identify and briefly describe reports that are available to a supervisor via their PC monitor. Attach samples of available display screenshots.

Cisco Response: The Cisco Agent Desktop (CAD) Supervisor Desktop provides the following real-time statistics displays:

Team/Skill Statistics Skill Summary Statistics Skill Agent Statistics Team Agent Statistics Team Agent State Statistics Agent vs. Team Statistics Agent Call Log Agent ACD State Log Agent Active Call Enterprise Data Call History

9.5.4 How often is data updated on supervisor monitor display?

Cisco Response: Supervisor Desktop data is updated in real-time for agent call activity. Real-time reports are refreshed every 10 seconds.

9.5.5 Describe the various display screens a supervisor would have access to for real time management operations. Include an explanation for each



screen field. Specify if any of the following agent performance metrics are displayed: service level; quantity and time of calls in queue; average speed to answer; number of agents staffed to handle calls; and identity any other available performance monitoring capabilities.

Cisco Response: The Cisco Agent Desktop (CAD) Supervisor Desktop is designed as a single cockpit to make it easy for the supervisor to quickly access the state of the team and contact center. There are not individual screens required to access data, the data can be pre-defined as statistics tabs and automatically displayed for the supervisor.

9.5.6 Can a supervisor perform drag and click system management programming from their monitor, and if so briefly describe available management features and operations. Specify if a supervisor can perform real time reconfiguration of call flows and agent skills assignments.

Cisco Response: The Cisco Agent Desktop (CAD) Supervisor Desktop allows the supervisor to click on their agents and quickly get information about the calls they are on, modify their state and initiate a silent monitoring, barge-in or intercept of an active call with that agent – all with a few clicks of the mouse. The desktop also provides a button for agent reskilling.

9.5.7 Can supervisors log-out an agent remotely?



Cisco Response: Yes.

9.5.8 Can supervisors force a group into night handling?


9.5.9 Can supervisors monitor and observe agents by agent ID? Can they listen and talk to an agent during a conversation: Can the entire customer experience be monitored, including announcements, music, etc.?

Cisco Response: Supervisors can monitor calls at the agents during a conversation.

9.6 Agent Functions

The proposed ACD solution must support a variety of agent functions. Provide answers to each of the following agent function requirements.

9.6.1 Describe the process for an agent to request supervisor assistance during an active call, and specify if the caller must be on hold or if the agent/supervisor can talk without the caller hearing the conversation. After the supervisor consultation can the caller be transferred or conferenced?

Cisco Response: The Cisco Contact Center Enterprise system provides for a button on the agent desktop to request assistance – this button triggers a pre-defined routing script that selects the supervisor to join the call. The system can be configured that the supervisor is joined via “warm conference” where the agent and supervisor will be able to speak while the caller is kept on-hold, or the supervisor can be joined directly via “blind conference.”

The caller can be on hold or actively speaking with the agent when the supervisor assist function is called – and there are two levels of assist built into the system to allow for “emergency” treatment that might invoke a recording of the call and send messages to security or others in the organization or a “assist” function that only brings in the supervisor as needed.

Because this functionality is implemented using standard call control functionality, there are no specific considerations of how the call can be handled once the supervisor joins – it acts like any other 3-way conference call in the system.

9.6.2 What happens if a supervisor is not available when assistance is required?

Cisco Response: If the supervisor assist script does not find an available supervisor, message is displayed to the agent on their desktop and the caller is kept in the call with the agent.



9.6.3 Identify and provide a brief description of ACD system information (calls in queue, average time in queue, personal call handling statistics, et al) that can be retrieved and displayed on the agent desktop telephone instrument.

Cisco Response: Using the Cisco Agent Desktop (CAD) IP Phone Agent function, the Cisco IP Phone XML display can be used to display call and contact center statistics:

For each skill group assigned to the agent, the number of calls in queue (CIQ) and the longest in queue (LIQ)

When calls arrive at this IP Phone, the phone can provide basic caller information as well, which can be pre-defined by agent team:

9.6.4 Identify and provide a brief description of ACD system information and reports (calls in queue, average time in queue, personal call handling statistics, et al) that can be retrieved and displayed on the agent PC monitor. Attach samples of available display screenshots.



Cisco Response: The Cisco Agent Desktop (CAD) can provide statistics to the agent about their own performance including: Agent State/Status Agent Call Log Agent ACD State Log Agent Detail Real-Time Skills Statstics Real-Time

9.6.5 Describe how an agent is able to distinguish incoming calls as a new call, transferred call, or a call from a voice response system.

Cisco Response: The Cisco Agent Desktop (CAD) provides the agent with a call appearance window that alerts them to the new incoming call, but also provides a historical track of where the call has been in the system—making it easy for agents to quickly understand how long a caller spent in the IVR, how long they were in queue or if they had spoken to other agents in this same interaction.

9.6.6 Describe each of the available “states” an agent can be in, e.g., logged in; available to take calls; after call work time; etc.

Cisco Response: The Cisco Contact Center Enterprise provides for the following agent states:



Logged in – the agent has connected to the system – they can also be either Ready or Not Ready while being logged in

Not Ready – the agent is logged in, but not available to take calls

Ready – the agent is logged in and available to take calls

Reserved – the agent has been selected to handle a call, the reserved state is used to associate the agent with the call being sent to them

Hold – the agent is logged in and has placed a call on hold in the system

After Call Work – (optional) the agent has completed an ACD call and is being held in a post-call state to allow them to enter wrap-up data about the call. This can be configured as “timed” to automatically put agents into this state for a pre-defined amount of time. Agents can exit this state by entering a Ready or Not Ready state.

9.6.7 Describe how an agent enters work codes that describe the nature of the call. Specify the maximum number of work codes and work code digits that can be entered into the system.

Cisco Response: Cisco Contact Center Enterprise allows for the configuration of “work codes” that agents can be prompted for at various state changes – these can be defined as optional, required or not allows as a system parameter or by agent settings:

Wrap-up – after a call is complete, the agent is presented a list of pre-defined codes to select from

Not Ready – after the agent selects the not ready state, the agent is presented with a list of pre-defined codes to select from

Logout – after the agent selects to log out of the system, the agent is presented with a list of pre-defined codes to select from

9.6.8 Can agents be made automatically available immediately after each call?


9.6.9 Can agents be members of multiple groups/splits/skills?


9.6.10 Can agents be made automatically unavailable after each call in order to complete work associated with the call before the next call is delivered? Can this time be specified and controlled and is this unavailable state measured and tracked in ACD reports?



Cisco Response: Yes, agents can be assigned “timed after call work” mode – which automatically puts them back into available state at the end of the timer value and the time in this state is tracked/reported by the system.

9.6.11 Can agents make themselves unavailable temporarily and have this unavailable state measured and tracked in ACD reports. Can the ACD agent enter a reason code to indicate why they are unavailable and have this unavailable state measured and tracked by reason code on ACD reports?

Cisco Response: Yes. Agents can select a Not Ready state and optionally can be prompted to enter a not-ready reason code. Both the time and codes are tracked for reporting purposes.

9.6.12 Can calls that ring at an available agent’s station, but are not answered automatically, be redirected to the next available agent rather than letting the call ring unanswered until abandonment? For example, if an agent left their station without logging out, will the system automatically log the agent out or make them unavailable and notify the supervisor? Will this event be tracked by the reporting system?

Cisco Response: Yes – the system automatically hands a “return no answer” call and automatically puts the agent into “not ready” state with a pre-defined reason code for tracking.

9.6.13 Can the system provide a brief announcement heard only by the agent indicating what type of call is arriving so that the agent can greet the caller appropriately if agents handle calls for multiple applications or who are visually impaired,? Can the voice terminal also display this information to the agent before delivery of the call?

Cisco Response: Yes, the system can be set up to provide a pre-recorded or dynamic “whisper” to the agent prior to call arrival. The same information is also made available on the agent desktop. For low vision agents, the screen can be made larger and screen reader software can also be used. The IP Phone XML display can also show caller information as well.

9.6.14 Can agents be “logical agents”, i.e., can they login with their agent ID from any system endpoint and take ACD calls?

Cisco Response: Yes, using the Cisco Unified Communications Manager extension mobility feature, agents can log into any phone with their agent ID to handle contacts.

9.6.15 Confirm if the system can automatically record agent calls for quality and monitoring purposes, and indicate if all calls can be recorded (incoming, station-to-station, non-ACD calls, etc.). Indicate if there is a beep tone to notify one or both of the call parties that the call is being recorded. Also indicate if the proposed ACD solution supports an integrated call recording feature or if an auxiliary system is required.

Cisco Response: The Cisco Agent Desktop (CAD) provides a manual call recording function for agents to be able to record specific calls. For more programmatic call recording, a 3rd party recording vendor like Witness/Verint or



NICE would be recommended to provide legal/regulatory recording functionality.

9.7 System Call Prompts & Announcements

VoiceCon requires that the proposed ACD solution prompt the caller to enter digits to determine how a call should be routed and then route based upon their response.

Vendor Response Requirement: Confirm if the proposed ACD solution has a fully integrated call prompt (call director) feature that does not require optional hardware/software equipment? If yes, identify how call prompt announcements are programmed, scripted and recorded; the maximum number of programmable call prompts; and the maximum number of concurrent activated call prompts for incoming calls. Also indicate if the caller can interrupt the call prompt for TUI response before the full script is played.

Cisco Response: The Cisco Unified Contact Center Enterprise using either IP-IVR or CVP can integrate with MRCP-based standard speech engines to provide automatic speech recognition (ASR) for caller prompts – the ASR functions are integrated into the Cisco scripting engine to simplify the creation and management of the prompting.

9.7.1 Hands Free Caller Prompt Response

Vendor Response Requirement: Confirm if the proposed ACD solution can support integrated Automatic Speech Recognition (ASR) as an alternative to TUI and describe how speech prompts are scripted and recorded.

Cisco Response: The Cisco Unified Contact Center Enterprise using either IP-IVR or CVP can integrate with MRCP-based standard speech engines to provide automatic speech recognition (ASR) for caller prompts – the ASR functions are integrated into the Cisco scripting engine to simplify the creation and management of the prompting.

9.7.2 Announcements

Vendor Response Requirement: Confirm that the proposed ACD solution has fully integrated announcement capabilities and provide answers to the following:

How many different announcements can be provided?

Cisco Response: The number of announcements is unlimited

How many announcements can be played concurrently?



Cisco Response: There is no limit to the number of announcements that can be played concurrently

How many announcement boards are supported?

Cisco Response: The Cisco solution does not use announcement boards to play messages to callers.

Are digitally recorded announcements supported?

Cisco Response: All announcements are stored as .WAV files in digital format.

How many different music sources can be supported?

Cisco Response: Typically, customers use pre-recorded/licensed music sources that are pre-recorded as .WAV files and made available on multiple media servers or make use of a network (RTPS) real time streaming media source for music in queue. Alternatively, music can be sourced from the Unified Communications Manager “hold” music source either from stored .WAV files or from a live music (or external music-on-hold server) source.

Can multiple announcements and music treatment be provided to a call, and can announcements and music treatment be specific to each queue?

Cisco Response: Yes, the Cisco solution treats each call individually, on a call-by-call basis and can select in the routing script the proper set of announcements, music treatment, and other prompting for each caller in queue – unlike traditional ACD systems which are limited to providing different treatment only by creating a different queue and all calls in that queue can only get the same treatment.

Can announcements and music treatment provided depend upon queue conditions or call related information, and how many different announcements can be provided for this situation?

Cisco Response: Yes, the Cisco solution can use real-time conditions to select the appropriate announcements and music treatments to play to callers on a call-by-call basis.

How do you handle feedback (music/announcements) for calls that are queued remotely?

Cisco Response: In the Cisco solution using Customer Voice Portal, callers are queued in the Cisco Voice Gateway at the remote site, using local .WAV files for messages and queue music that is cached in the gateway or provided from media servers on the network. Calls are not extended across the network to be put in queue, nor is queue music streamed across the network wasting bandwidth.

Can you connect audible feedback locally for calls that are queued remotely in order to decrease the number of packets sent over the IP trunk?

Cisco Response: Yes, the design of the Cisco solution with Customer Voice Portal for queuing and treatment is to reduce the need to send packets across the network for messaging and queuing – keeping the caller at the edge of the network until there is an available agent to handle the call.



9.8 MIS Reporting System

VOICECON requires a comprehensive MIS reporting system capability with its proposed ACD solution.

In 2008, Cisco introduced the new Cisco Unified Intelligence Suite (CUIS) web-based reporting tool as an optional enhanced analytics and dashboard application for reporting. Cisco Unified Intelligence Suite allows Cisco Contact Center Enterprise data to be combined with external data sources (CRM, billing, 3rd party ACD/IVR, etc.) to provide actionable business intelligence for the contact center. CUIS is a separately priced option and offered in addition to the standard Cisco Contact Center Enterprise WebView web-based reporting package that is included free of charge with the ACD solution. For the sections that follow, both CUIS and WebView reporting tools are referenced in response to the specific vendor requirements.

Vendor Response Requirement: Confirm that the proposed ACD MIS reporting system supports each of the following capabilities and attributes, and identify any that are not supported:

Track local and remote, digital and IP agents

Cisco Response: All agents in the Cisco solution are tracked by the system – with the Cisco WebView and Cisco Unified Intelligence Suiteweb-based reporting tools as well as at the local site Supervisor Desktop application.

Windows-based GUI

Cisco Response: The Cisco Contact Center Enterprise platform provides reporting using a web-based GUI tool to review and create reports, there is no Windows-based reporting client required.

Real-Time Monitoring

Cisco Response: The Cisco WebView and CUIS reporting tools provides real-time and historical reports in both tabular and graphical (graphs) formats over a secure (HTTPS) web connection. CUIS provides additional “dash board” functionality to allow users to easily create their own custom dashboards of real-time data across multiple sources of data, including external web-services based widgets. The Supervisor Desktop also provides real-time tabular and graphical reports as well for the agent teams.

Reporting Exceptions

Cisco Response: The Cisco WebView and CUIS reporting tools allows users to define specific thresholds on reports that allow the data elements to be highlighted on the screen in real-time to call out exceptions. CUIS allows for additional actions to be taken based on the thresholds, such as calling an external web service to notify the supervisor or manager of the exception.

Threshold Notification

Cisco Response: The Supervisor Desktop provides for “supervisor workflows” that are triggered on real-time conditions such as the number of calls in queue, queue time, number of agents logged in, etc. These workflows can send



automatic notifications to the supervisors to alert them to conditions that require attention in their contact centers. Likewise, CUIS can use these thresholds to trigger an external action such as a web service call for notification.

Web-based access to reports

Cisco Response: The Cisco WebView and CUIS reporting tools provide secure, web-based tools for reporting.

Historical Reporting

Cisco Response: The Cisco WebView and CUIS reporting tools provides historical reports.

Custom Reporting Option

Cisco Response: The Cisco WebView reporting tool can be extended to include custom reports created by the users/administrators using the Sybase Infomaker tool. The Sybase tool is used to create the standard WebView report templates, users can modify the standard templates or create new ones that join data from the different tables in the open Cisco database. Cisco publishes the schema for the contact center database, allowing for 3rd party tools to be used as well for additional extract/data analysis. The Cisco Unified Intelligence Suite allows for easy creation of custom reports and dashboards using pre-built wizards to guide the user in the creation of the report/dashboard as well as being able to copy and modify existing views into the system, without having to use a 3rd party tool outside of the CUIS application. (This is one of the major advantages CUIS provides to customers over the standard WebView reporting tool)

Open Database Connectivity

Cisco Response: The Cisco solution uses the Microsoft SQL Server database with a fully published schema. Users and administrators can connect to the databases using ODBC to perform their own custom queries as needed.

Exporting Data to other applications

Cisco Response: The Cisco solution allows for the data in the Microsoft SQL Server database to be exported to external applications and customer data warehouse applications or 3rd party applications like work force management.

Local and remote access by supervisors

Cisco Response: The Cisco WebView reporting tool can be accessed locally or remotely by users as needed—there is no restriction in the system.

9.8.1 System Requirements

Vendor Response Requirement: Briefly describe the hardware/software requirements for the proposed ACD MIS reporting system, including requirements for applications servers and software, workstation terminals (agent and supervisor) and client software.

Cisco Response: Both the Cisco WebView and CUIS reporting tools runs on a standard Windows/Intel platform using a Microsoft Windows IIS web server to



provide secure, web-based reports to browser client users. The typical WebView or CUIS Application/Database server would be configured as follows:

Cisco MCS-7845 Server (H2) Xeon 3.22GHz dual processor with 4GB RAM Microsoft Windows Server 2003 Standard Edition, SP1 and/or R2 or SP2 Microsoft SQL Server 2000 Standard Edition, SP4.

The Cisco Unified Intelligence Suite has two components: CUIS Application Server and the CUIS Database Archiver. These two components can be co-loaded on the same physical server (as described above) or broken out for additional concurrent users and/or database retention/sizing.

9.8.2 MIS Reports

The ACD MIS reporting system must be able to support a wide variety of report categories, including, as a minimum, the following:

Single Agent Reports Agent Group Reports ACD Queue Reports Abandoned Calls Report Trunk Reports Daily Total Reports Ongoing Status Reports


Confirm the proposed fully integrated ACD MIS reporting system satisfies this requirement, and identify any listed report type not supported.

Cisco Response: The Cisco WebView reporting tool can provide all of these types of reports including agent level detail (state trace and reason code reporting). The reporting categories include:



agent

application gateway (external data access)

application path (multi-channel)

outbound option

call type

peripheral

route

routing client

script queue

service

skill group

trunk group

The Cisco Unified Intelligence Suite provides detail real-time and historical report templates in the following categories as well:



Agent Reports

Call Type Reports

Service Reports

Skill Group Reports

Trunk Group/IVR Report



9.8.3 Real Time Reports


Provide a list of all standard and optional real time monitoring reports.

Cisco Response: Unified Contact Center Enterprise (UCCE) WebView reporting tool incluees Report Templates: Agent real-time reports agent20: Agent Real Time Report agent28: Agent Real Time All Fields Report Agent by Peripheral Reports Agent by Peripheral Real-Time Reports agtper20: Agent Peripheral Real Time Report agtper28: Agent Peripheral Real Time All Fields Report Agent by Peripheral Historical Reports agtper03: Agent Peripheral Media Logout Status Report agtper04: Agent Peripheral Task Detail Activity Report agtper05: Agent Peripheral Task Detail Performance Report agtper21: Agent Peripheral Task Summary Half Hour Report agtper22: Agent Peripheral Task Summary Daily Report agtper23: Agent Peripheral Performance Summary Half Hour Report agtper24: Agent Peripheral Performance Summary Daily Report agtper25: Agent Peripheral Consolidated Half Hour Report Template agtper26: Agent Peripheral Consolidated Daily Report agtper27: Agent Peripheral Historical All Fields Report Agent By Skill Group Reports Agent by Skill Group Real-Time Reports agtskg06: Outbound Option (Blended Agent) Status Report agtskg28: Agent Skill Group Real Time All Fields Report agtskg30: UCCE Agent Skill Group Real Time Report Agent By Team Reports Agent by Team Real-Time Reports agteam02: Agent Skill Group Status Report agteam20: Agent Team Real Time Report agteam28: Agent Team Real Time All Fields Report agteam29: Agent SkillGroup Assignments Real-Time agteam32: Agent Team State Counts Real Time Report UCCE CallType Reports UCCE Call Type Real-Time Reports caltyp04: Call Type Service Levels Real Time Report caltyp26: Call Type Tasks Offered Over Half Hour caltyp27: Call Type Queue Delay Status Real Time



caltyp28: Call Type Task Status Now Real Time Report (UCCE and Network VRU) UCCE Peripheral Service Report Templates persvc27: Peripheral Service Real Time All Fields Report UCCE Skill Group Report Templates UCCE Enterprise Skill Group Real-Time Reports entskg01: Enterprise Skill Group Status Real Time Report Template entskg03: Enterprise Skill Group Agent Status Report entskg05: Enterprise Skill Group % Utilization of Ready Agents Report caltyp20: Call Type Real Time Report caltyp24: Call Type Real Time All Fields Report caltyp25: Call Type Queue Status Real Time Report (UCCE and Network Queue) entskg14: UCCE Rolling 5-Minute Enterprise Skill Group Status Report entskg28: Enterprise Skill Group Real Time All Fields Report entskg29: Enterprise Skill Group Logout Real Time Report entskg30: UCCE Enterprise Skill Group Status Real Time Report UCCE Peripheral Skill Group Reports perskg01: Peripheral Skill Group Status Real Time Report perskg03: Peripheral Skill Group Agent State Status Report perskg05: Peripheral Skill Group % Utilization of Ready Agents Report perskg11: Outbound Option (Blended Agent) Statistics By Skill Group Report perskg14: UCCE Rolling 5-minute Peripheral Skill Group Status Report perskg28: Peripheral Skill Group Real Time All Fields Report perskg29: Peripheral Skill Group Logout Real Time Report perskg30: UCCE Peripheral Skill Group Status Real Time Report Trunk group for IP-IVR reports trkgrp04: Trunks Real Time All Fields Report trkgrp20: All Ports Busy Real Time Report trkgrp21: IVR Ports Idle & In Service Real Time Report trkgrp22: IVR Ports Status Real Time Report Application Gateway, Path, and Script Queue Reports appath01: Application Path Real Time Report scrque01: Script Queue Node Real Time Report Outbound Option (Blended Agent) Reports camqry02: Summary of Call Counts Per Campaign Real Time Report camqry03: Valid Campaign Dialing Times Real Time Report camqry04: Query Rule Dialing Times Real Time Report camqry05: Call Summary Count Of Query Rule Within Campaign Real Time camqry06: Call Summary Count per Campaign Real Time dialer01: Dialer Real Time Report dialpr01: Dialer Port Status Real Time Report imprul01: Import Status Real Time Report The Cisco Unified Intelligence Suite provides real-time report templates that can be used to create any number of new/customized reports and/or



dashboards within the CUIS application. The “stock” templates include all relevant fields stored in the Cisco Contact Center Enterprise database:

Agent Reports Agent Real Time All Fields Agent Not Ready Detail Agent Team Real Time All Fields Agent Team State Counts Real Time

Call Type Reports Call Type Real Time All Fields

Service Reports Peripheral Service Real Time All Fields

Skill Group Reports Agent Skill Group Real Time All Fields Enterprise Skill Group Real Time All Fields Peripheral Skill Group Real Time All Fields

9.8.4 Historical Reports


Provide a list of standard and optional historical system reports.

Cisco Response: Unified Contact Center Enterprise (UCCE) WebView Reporting Tool includes Report Templates: Agent by Peripheral Historical Reports agtper03: Agent Peripheral Media Logout Status Report agtper04: Agent Peripheral Task Detail Activity Report agtper05: Agent Peripheral Task Detail Performance Report agtper21: Agent Peripheral Task Summary Half Hour Report agtper22: Agent Peripheral Task Summary Daily Report agtper23: Agent Peripheral Performance Summary Half Hour Report agtper24: Agent Peripheral Performance Summary Daily Report agtper25: Agent Peripheral Consolidated Half Hour Report Template agtper26: Agent Peripheral Consolidated Daily Report agtper27: Agent Peripheral Historical All Fields Report Agent by Skill Group Historical Reports agtskg03: Agent Skill Group Logout Status Report agtskg04: Agent Task Detail Activity Report agtskg05: Agent Task Detail Performance Report agtskg07: Agent Skill Group Task Analysis Report agtskg10: Outbound Option (Blended Agent) Predictive and Progressive Tasks Detail Performance Report



agtskg11: Outbound Option (Blended Agent) Preview Task Detail Performance Report agtskg12: Outbound Option (Blended Agent) Reservation Task Detail Performance Report agtskg21: Agent Skill Group Task Summary Half Hour Report agtskg22: Agent Skill Group Task Summary Daily Report agtskg23: Agent Skill Group Performance Summary Half Hour Report agtskg24: Agent Skill Group Performance Summary Daily Report agtskg25: Agent Skill Group Consolidated Half Hour Report agtskg26: Agent Skill Group Consolidated Daily Report agtskg27: Agent Skill Group Historical All Fields Report Agent by Team Historical Reports agteam03: Agent Logout Status By Team Report agteam04: Agent Task Detail Activity Report agteam05: Agent Task Detail Performance Report By Team agteam21: Agent Team Task Summary Half Hour Report agteam22: Agent Team Task Summary Daily Report agteam23: Agent Team Performance Summary Half Hour Report agteam24: Agent Team Performance Summary Daily Report agteam25: Agent Team Consolidated Half Hour Report agteam26: Agent Team Consolidated Daily Report agteam27: Agent Team Historical All Fields Report agteam33: Agent Team Incoming/Outgoing Task Durations With Agent Detail Half Hour agteam34: Agent Team Incoming/Outgoing Task Durations With Agent Detail Daily agteam35: Agent Team Incoming/Outgoing Task Durations Half Hour agteam36: Agent Team Incoming/Outgoing Task Durations Daily UCCE CallType Reports UCCE Call Type Historical Reports caltyp05: Analysis of Calls Half Hour Report caltyp21: Call Type Half Hour Report caltyp22: Call Type Daily Report caltyp23: Call Type Historical All Fields Report caltyp31: Call Type Abandon/Answer Distribution by Half Hour Report caltyp32: Call Type Abandon/Answer Distribution Report caltyp33: Call Type Abandon/Answer Cumulative Distribution by Half Hour Report caltyp34: Call Type Abandon/Answer Cumulative Distribution Report caltyp35: VRU Calls Analysis Half Hour Report caltyp36: VRU Calls Analysis Daily Report caltyp37: Call Type Service Level Abandons Daily Report UCCE Peripheral Service Report Templates UCCE Peripheral Service Reports persvc20: Peripheral Service for IVR Queue Half Hour Report persvc21: Peripheral Service IVR Queue Daily Report persvc22: Peripheral Service IVR Self-Service Half Hour Report persvc23: Peripheral Service IVR Self-Service Daily Report persvc24: Peripheral Service Agent Half Hour Report persvc25: Peripheral Service Agent Daily Report



persvc26: Peripheral Service Historical All Fields Report UCCE Skill Group Report Templates UCCE Enterprise Skill Group Historical Reports entskg06: Enterprise Skill Group Performance Half Hour Report entskg07: Enterprise Skill Group Performance Daily Report entskg08: Full Time Equivalent for Enterprise Skill Groups Half Hour Report entskg09: Enterprise Skill Group Normalized Agent State Report entskg27: Enterprise Skill Group Historical All Fields Report entskg31: UCCE Enterprise Skill Group Task Summary Half Hour Report entskg32: UCCE Enterprise Skill Group Task Summary Daily Report entskg33: UCCE Enterprise Skill Group Performance Summary Half Hour Report entskg34: UCCE Enterprise Skill Group Performance Summary Daily Report entskg35: UCCE Enterprise Skill Group Consolidated Half Hour Report entskg36: UCCE Enterprise Skill Group Consolidated Daily Report UCCE Peripheral Skill Group Reports Peripheral Skill Group Historical Reports perskg08: FTE for Peripheral Skill Groups Half Hour Report perskg09: Peripheral Skill Group Normalized Agent State Report perskg12: Outbound Option (Blended Agent) Task Detail Performance In Skill Groups Half Hour Report perskg27: Peripheral Skill Group Historical All Fields Report perskg31: UCCE Peripheral Skill Group Task Summary Half Hour Report perskg32: UCCE Peripheral Skill Group Task Summary Daily Report perskg33: UCCE Peripheral Skill Group Performance Summary Half Hour Report perskg34: UCCE Peripheral Skill Group Performance Summary Daily Report perskg35: UCCE Peripheral Skill Group Consolidated Half Hour Report perskg36: UCCE Peripheral Skill Group Consolidated Daily Report Trunk group for IP-IVR reports trkgrp12: Trunks Historical All Fields Report trkgrp23: IVR Ports Performance Half Hour Report Application Gateway, Path, and Script Queue Reports apgate11: Application Gateway Status Half Hour Report scrque01: Script Queue Node Real Time Report Outbound Option (Blended Agent) Reports Outbound Option Historical Reports camqry10: Call Counts of Query Rule within Campaign Half Hour Report camqry11: Summary of Call Counts per Campaign Half Hour Report dialer10: Dialer Call Result Summary Half Hour Report imprul10: Import Rule Report The Cisco Unified Intelligence Suite provides historical report templates that can be used to create any number of new/customized reports and/or dashboards within the CUIS application. The “stock” templates include all relevant fields stored in the Cisco Contact Center Enterprise database:



Agent Reports Agent Historical All Fields Agent Not Ready Detail Agent Team Historical All Fields

Call Type Reports Call Type Abandon/Answer Distribution Historical Call Type Historical /Call Type Daily All Fields

Service Reports Enterprise Service Historical All Fields Peripheral Service Historical All Fields

Skill Group Reports Agent Skill Group Historical All Fields Enterprise Skill Group Historical All Fields Peripheral Skill Group Historical All Fields

Trunk Group/IVR Report IVR Ports Performance Historical

9.8.4.1 Frequency


What is the frequency that reports can be produced?

Cisco Response: All reports in the Cisco WebView and CUIS solution can be produced “on demand” or scheduled to run using a “job scheduler” on a daily, weekly, monthly basis at specific times/dates. Reports can be scheduled to print at a network printer, write to a specific network share, or for CUIS reports publish an RSS feed automatically.

9.8.4.2 On-Demand Reporting


Can all of the historical reports be accessed on-demand? Identify any report that cannot be accessed on-demand.

Cisco Response: All reports in the Cisco solution can be requested “on-demand” however, historical half-hour reports are only provided for prior half-hour periods.

9.8.4.3 Storage & Backup




How long are historical reports stored and archived by the system, and describe data backup operations.

Cisco Response: Data in the Cisco solution is kept in full detail. The amount of historical data that can be kept on the system is only limited by available disk space. The Cisco Unified Intelligence Suite Database Archiver can be set up to automatically produce summary daily, weekly, monthly, etc. tables to speed query and database access for historical reporting as well.

9.8.5 Customized Reporting

VoiceCon may require customized ACD MIS reports in addition to system standard/optional reports.


Confirm the proposed ACD MIS reporting system supports customized reporting capabilities and provide a brief description of how customized reports can be defined and generated.

Cisco Response: Yes. The report templates used by the Cisco WebView reporting tool are created using Sybase Infomaker. Users may use the Infomaker tool to modify the existing report templates and/or create new report templates that draw data from the fully published database schema. User created/modified templates are loaded directly into the WebView server and can be used by any other user on the platform or restricted as “personal” reports.

The Cisco Unified Intelligence Suite also allows for easy creation of custom reports and dashboards using pre-built wizards to guide the user in the creation of the report/dashboard as well as being able to copy and modify existing views into the system, without having to use a 3rd party tool outside of the CUIS application. (This is one of the major advantages CUIS provides to customers over the standard WebView reporting tool)

9.8.6 Terminal Report Access

VoiceCon requires that ACD MIS reports must be available on terminal display and paper printout and be able to be downloaded to a PC.


Confirm the proposed ACD MIS reporting system supports this requirement.

Cisco Response: Yes. The Cisco WebView and CUIS reporting tools present data to the user in their web browser as well as printed locally to any printer the user has access to. The report can also be downloaded to their local PC in a



number of common formats: PDF, CSV (comma-separated values), DBF (dBase-II/III format), DIF (Data Interchange Format), Micrsoft Excel, HTML Table, Powersoft Report Format, SQL Insert, SYLK, Text, WKS, WMF).

9.8.7 Scheduled and Email of Reports

VoiceCon requires that ACD MIS reports must have email capability and permit the supervisor to schedule.


Confirm the proposed ACD MIS reporting system supports this requirement.

Cisco Response: The Cisco WebView reporting tool allows supervisors and administrators to schedule reports and target them to be printed at specific network printer locations. This same functionality can be used to automate e-mail distribution of reports. CUIS adds the ability to directly send reports via e-mail to users as well as to “publish” RSS feeds based on dashboards and reports automatically to users who have subscribed to the RSS service in CUIS.

9.8.8 Report Formats

VoiceCon requires that the proposed ACD MIS reporting system supports a variety of graphical and file type (EXCEL, TEXT, PDF, et al.) report formats.


Confirm the proposed ACD MIS reporting system supports a variety of graphical report formats and identify by type.

Cisco Response: Reports can be generated in a number of common formats: PDF, CSV (comma-separated values), DBF (dBase-II/III format), DIF (Data Interchange Format), Micrsoft Excel, HTML Table, Powersoft Report Format, SQL Insert, SYLK, Text, WKS, WMF).

9.9. ACD Management & Administration


Provide a description of ACD management & administration capabilities. Identify in your response which of the following supervisor or administrator capabilities are supported:

Define service levels and other thresholds Cisco Response: In the Cisco solution, administrators set up service level thresholds (percent of calls that must be answered within a service level target) using the configuration tools of the Unified Contact Center Enterprise. These are set at the system level (globally) as a default, which can be overridden at



each Service or Call Type in the system by the administrator using the configuration tools.

Thresholds on WebView reports can be set by the user who is viewing the reports and saved as a “favorite” report for the user or shared with other users in the system.

Thresholds on the agent and supervisor desktop are set by the system administrator in the Cisco Agent Desktop (CAD) Administrator configuration tool.

Create call flow scripts Cisco Response: Call flow scripts are created by the Administrator using the Script Editor tool.

Enter agent PINs and create passwords Cisco Response: Agents can be added and their PINs/passwords and other data modified by the Administrator or the Supervisor using the configuration tools of the system. Agents can use the Cisco Unified Contact Center Management Portal (UCCCMP) to change their passwords in the system directly.

Limit access to data for users Cisco Response: The Administrator can create “Feature Control Lists” using the configuration tools to limit the access a user has in the system to view/edit call flow scripts, data on reports and other features of the system. Additionally, the Cisco Unified Intelligence Suite (CUIS) provides a user-defined security model that can be used to associate specific users or groups of users to specific data elements – for

Configure all peripherals (printers, faxes etc.) Cisco Response: The Administrator configures peripherals in the system using the configuration tools.

Backup and restore the database Cisco Response: The Administrator can use Microsoft SQL Server Enterprise Manager tools to backup and/or restore the system database or their own 3rd party backup tools as needed.

Configure automatic backup and recovery of customer data Cisco Response: There is no automatic backup functionality included with the system, all of the critical data is stored in redundant databases that the system automatically keeps synchronized on redundant physical servers.

Create individual ”views” of the call-center Cisco Response: The Administrator associates supervisors with specific agent teams to create their own “view” of the call center – which is used by the Supervisor Desktop to limit the view of the supervisor to his/her assigned team as well as the WebView reporting tool to allow the supervisor to only view agent reports for his/her specific teams.



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 budgetary purposes, installation costs have been estimated at 15% of list price; the exact installation costs will vary, depending upon the Cisco Partner involved in the design and installation of the system, the exact location, additional site-specific work that may be required, etc. A conservative discount of 40% was 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.

Special Note: Cisco Unified Workspace Licensing (CUWL): Cisco Unified Workspace Licensing has been used in this proposal to provide VoiceCon with an easy and affordable program to procure a broad range of Cisco Unified Communications applications and services. Workspace Licensing, inclusive of all client and server software, licensing, service and support, software subscription for applications and clients, facilitates consistent deployment of multiple applications to all users in their workspaces, and helps organizations maximize the potential of unified communications.

This program streamlines pricing, licensing, software subscription and acquisition of Cisco Unified Communications solutions and introduces the ability for businesses, government agencies, and institutions to implement a media-rich unified communications experience at a lower per-user price point. Presence, unified clients, mobility, unified messaging, and audio, video, Web conferencing are just a few of the applications included in this program. What is Cisco Unified Workspace Licensing?

Cisco Unified Workspace Licensing offers two options, Standard Edition and Professional Edition. The following table compares the two options. Note: The



Professional Edition was used in this proposal for VoiceCon. Pricing details are discussed in Part 2 “System Pricing”. Please visit the following web site for more information: http://www.cisco.com/go/workspace_licensing

Functionality What’s Included in the Workspace License

Standard Edition

Professional Edition

Video Conferencing

Cisco Unified MeetingPlace Express (25 Cisco UWL=1 Port)

N Y

Web Conferencing


N Y

Audio Conferencing


N Y

Mobile Phone Client

Cisco Unified Mobile Communicator Client

N Y

Contact Center


N Y

Presence Cisco Unified Presence Profile

Y Y

Mobility (with Sim Ring services)

Cisco Unified Mobility Profile Y Y

Microsoft MOC Integration

Cisco Unified Communication Integration with Microsoft

Y Y

Soft Client Cisco Unified Personal Communicator or Cisco Unified IP Communicator with Cisco Unified Video Advantage

Y Y

Messaging Cisco Unity or Cisco Unity Connection

VM VM/UM

Phone/Call Control

License for One or Unlimited Cisco IP Phones per User

One Unlimited



Three elements are priced in for each Unified Communication User:

Cisco Part Number

Description What’s Included Per User Price

LIC-UWL-PRO Unified Workspace Licensing Professional Edition – 1 User

Includes client and server licenses for all the Unified Communications applications listed above (Unified Communications Manager, MeetingPlace, Unity, Mobility, and Presence)

$425

UCSS-UWL-PRO Unified Communications Software Subscription contract – 3 years

Three years of client and server software upgrades

$125

CON-ESW-CUWLPRO1

Essential Operate Services contract

Provides maintenance and minor updates on Unified Communications software applications

$26

Analog Phones and Public Space Devices: IP Phones not associated to an individual such as conference room phones, lobby phones, wall phones, break room phones and other public space devices may be added to a Cisco Unified Workspace Licensing order. Likewise, analog phones or other analog devices such as fax machines, credit card scanners and other such devices that do not require any applications may be added as well. In both cases only the right to use for the corresponding phone will be provided and no applications (e.g. voicemail, presence, conferencing or any other applications) may be used with these phones. Public space IP phones may not exceed 15% of the total devices utilized on a workspace licensing cluster, and total public space devices (analog + IP) may not exceed 50%. Analog phones and public space devices are considered users under Cisco Unified Workspace Licensing, and so require Cisco Unified Communications Software Subscription. The table below summarizes CUWL pricing for these users/devices. Product Number

Description List Price (US$)

UCSS Product Number (3-Year)

UCSS List Price -3-year (US$)

Essential Operate Services Product Number

Service List Price – per year (US$)

ANLG-

DEV-UWL

Analog, non-app device add-on for UWL

$50 UCSS-

ANLGUWL-3-1

$15 CON-

ESW-

DEVUWL

0$

PUBLIC-

IP-DEV-

UWL

Public Space non-app phone add-on for UWL for lobby and conference room phones

$150 UCSS-

PUBUWL-3-1

$45 CON-

ESW-

IPDEVUWL

4$



2.0 Summary Pricing – VoiceCon IPTS solution Complete and submit the following table for your proposed IPTS solution pricing summary data in an attached file using EXCEL. The submitted data will be made available to the general public.

Component LIST DISCOUNT

All Common Equipment 285,370 171,222 Standard Generic Software License * 738,095 442,857 Optional Software Licenses (including, if applicable IP port license fees, all soft client license fees, mobile license fees) ** 30,995 18,597 Voice Terminal Equipment 1,210,849 726,509 Systems Management *** 0 0 Voice Messaging **** 23,000 13,800 Installation/Warranty fees ***** 736,007 736,007 TOTAL 3,024,316 2,108,993

* Includes all server and client licenses. See discussion of Cisco Unified

Workspace Licensing above. ** Cisco Emergency Responder application and hardware. *** Included with Cisco Unified Communications Manager. **** Hardware only, to support Cisco Unity (Unified Messaging). Server and

Client Licences are included in Cisco Unified Workspace License, so they are part of line two, “Standard Generic Software”

***** Includes estimated installation charges, plus Unified Communications

Software Subscription (3 years), plus Unified Communications Essential Operate Services (1 year)



2.2 Software Upgrade Subscriptions

Complete and submit the following software upgrade subscription pricing table for your proposed IPTS solution in an attached EXCEL file. The submitted data will be made available to the general public.

Software Element List Discount Annual Standard Generic Software Upgrade Subscription Fee (per user) *

125 * 75 *

Annual Voice Mail Software Upgrade Subscription Fee (per user)

Included in fee above

Included in fee above

* This is for a 3-year Unified Communications Software Subscription (UCSS),

thus the annual cost is $25 per user (discounted). UCSS includes all Cisco Unified Application server and client licenses that the customer orders under Cisco Unified Workspace Licensing. Analog devices and public IP phones (such as, conference phones) are charged a separate, lower fee. See discussion of Cisco Unified Workspace Licensing above.



3.0 Desktop Voice Terminal Pricing Complete and submit the following table for your proposed unit pricing for voice terminal equipment in an attached file using EXCEL. The submitted data will be made available to the general public. Terminal Cisco Product List Discount Economy Desktop Telephone


175.00 105.00

Administrative Desktop Telephone

Cisco Unified IP Phone 7962G plus IP Phone Expansion Module 7915

900.00 (505.00 +

395.00)

540.00

Professional Desktop Telephone


595.00 357.00

Executive Desktop Telephone


705.00 423.00

Attendant Soft Console

Arc Attendant Console 4,500.00 2,700.00

IP Audio Conferencing Unit

Cisco Unified IP Phone 7937 1,295.00 777.00

IP Terminal License Fee

*Included with CUWL

Mobile Extension License Fee

*Included with CUWL

Advanced Mobile Client License Fee

*Included with CUWL

Desktop Telephone Options:

Gigabit Ethernet Adapter

N/A – Gigabit ports built into phones.

Bluetooth adapter N/A DECT adapter N/A Key Modules Cisco Unified IP Phone

Expansion Module 7915 395.00 237.00

* Refer to Cisco Unified Workspace Licensing (CUWL) discussion above.



4.0 Detailed Pricing Summary Submit a separate EXCEL file with a detailed listing of proposed communications system components/elements and associated unit pricing. If possible 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 (cabinet or server equipped with processor, memory, power, and interfaces)

- 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 in support of required features and functions

- All voice messaging system elements, including cabinet equipment and memory storage

- All systems management elements The detailed pricing will NOT be made public, but will be used to verify adherance to system configuration performance requirements and the pricing summary data.

Cisco Response: A detailed Excel spreadsheet was submitted with this RFP Response.



Appendix A – Cisco Unified Communications Manager Release 7.0

Product Overview - Cisco Unified Communications Manager is the powerful call-processing component of the Cisco Unified Communications Solution. It is a scalable, distributable, and highly available enterprise IP telephony call-processing solution.

New with Cisco Unified Communications Version 7.0

Features Highlights and Benefits

As user needs evolve, Cisco Unified Communications Manager continues to evolve to meet those needs. Cisco Unified Communications Manager Version 7.0 aims to lower the total cost of ownership for organizations and improve the calling experience for end users as well as system administrators. Some of the key features of the recent release follow: • Local route groups and transformation patterns greatly reduce the configuration effort to create dial plans.

• Intelligent bridge selection saves you money by optimizing the use of video bridge resources.

• Trusted relay points facilitate trusted quality of service (QoS) and Call Admission Control (CAC), as well as trusted VLAN traversal for Cisco Unified Communications software clients.

• Early offer support with support of G.729 on Session Initiation Protocol (SIP) trunks provides the savings of low-bandwidth codecs

A complete list of new features included in Cisco Unified Communications Manager Version 7.0 is presented in Table 1.

Table 1. New Features in Cisco Unified Communications Manager 7.0

For Easier Administration, Saving You Time and Resources

• Calling party normalization

• E.164 with " + " dialing

• Local route groups and transformation

• Trusted relay point

• Intelligent bridge selection

Mobility Features

• Dial via Office

• Directed call park

• Reverse callback

• Simultaneous ring time-of-day access list

Greater Interoperability with Partners

• Click to conference with IBM Sametime

• Simultaneous ring Uniform Resource Identifier (URI) dialing with Microsoft OCS



• T.38 Fax interoperability with Microsoft Exchange

• Active Directory 2008

New Telephony Features

• Directed call pick up

• Do not disturb - call reject

• Directed call pickup

• Extension mobility feature safe

• Phone services provisioning

Additional Localization

• Estonian

• Latvian

• Lithuanian

New SIP Support

• Single button barge

• Join across lines

• Busy-lamp-field (BLF) alert

• BLF pickup

• Conference chaining

• Do not disturb - call reject

• Cisco Unified IP Phone 7931G

• Cisco Unified IP Phone Expansion Module 7914

• Secure Real-Time Transport Protocol (SRTP) over SIP trunk

• Early offer SIP trunk with G.729 with Media Termination Point (MTP)

• SIP trunk Preferred Asserted Identity (PAI)

Features for the Department of Defense (DoD)

• Assured Services Session Initiation Protocol (AS-SIP), Voice over Session Initiation Protocol VoSIP/DVX G.Clear

• Secure indication tone (Norway)

Added Support for Cisco Products

• Voice gateways: Cisco VG202 and VG204 Analog Voice Gateway models

• Click to dial on WebEx® meeting applications



• Cisco Emergency Responder Location Management user interface

• Cisco Security Agent 5.2 support

Direct Upgrades

• Cisco Unified CallManager 4.1(3) and 4.2(3) and Cisco Unified Communications Manager 4.3(1) and 4.3(2)

• Cisco Unified Communications Manager 5.1(3), 6.1(1), and 6.1(2)

Serviceability Enhancements

• Iptables

• Fresh install of Cisco Unified Communications Manager on the Cisco MCS 7828 Media Convergence Server

• Alerting subsystem in Cisco Unified Communications Operating System

Performance Improvements

• Reduce tracefile output by compression

• Database replication improvements

Product Specifications

Platforms

• Cisco MCS 7800 Series Media Convergence Servers, including Cisco MCS 7815, MCS 7816, MCS 7825, MCS 7828, MCS 7835, and MCS 7845

• Selected third-party servers; for details, visit http://www.cisco.com/go/swonly

• Cisco Unified Communications Manager Business Edition; for details, visit http://www.cisco.com/en/US/products/ps7273/index.html

The appliance model provides a platform for call processing with the software preloaded on a Cisco MCS platform; the software is optionally available as a DVD kit for equivalent customer-provided servers. The appliance comes with a single firmware image that includes the underlying operating system as well as the Cisco Unified Communications Manager application. The appliance is accessed through a GUI, and a command-line interface (CLI) has been added to facilitate diagnostics and basic system management such as the starting or stopping of services and rebooting of the appliance. No access to the underlying operating system is necessary. All system management activities, for example, disk space monitoring, system monitoring, and upgrades, are controlled through the GUI. Because onboard agents are no longer supported on the appliance, all Cisco Unified Communications Manager management interfaces are enhanced to allow tight integration with third-party applications. Additionally, the Simple Network Management Protocol (SNMP) interface has added an overall syslog performance MIB. The Serviceability interface has instrumented appliance-specific counters. The Programming interface has added the capability to run insert, update, and delete database commands. To further enhance security, Cisco Security Agent for Cisco Unified Communications Manager comes preloaded on the appliance.

Bundled Software

• Cisco Unified Communications Manager Version 7.0, a call-processing and call-control application, is included.



• The Cisco Unified Communications Manager Version 7.0 configuration database contains system and device configuration information, including the dial plan.

• Cisco Unified Communications Manager administration software is included.

• Cisco Unified Mobility is included.

• The Cisco Unified Communications Manager CDR Analysis and Reporting Tool (CAR) 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 tool also provides limited database administration - for example, deletion of records based on database size.

• The Cisco Unified Communications Manager Bulk Administration Tool (BAT) allows administrators to perform bulk add, delete, and update operations for devices and users. The application was enhanced in Version 6 to provide export and import of database information, including calling search space, device pool, and Cisco Survivable Remote Site Telephony (SRST). Version 7.0 further enhances the solution by adding many other features, among them hunt and pilot lists, computer telephony integration (CTI) route groups, transformation patterns, presence groups, message waiting, and mobility information.

• The Cisco Unified Communications Manager Attendant Console application is no longer bundled with Cisco Unified Communications Manager. It is, however, supported in this release for customers who have upgraded from a previous version.

• The Cisco Unified Communications Manager Real-Time Monitoring Tool (RTMT) monitors real-time behavior of the components in a Cisco Unified Communications Manager cluster. Cisco Unified Communications Manager RTMT uses HTTP and TCP to monitor device status, system performance, device discovery, and CTI applications. It also provides trace and log file management capabilities, including scheduling of download 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 Communications Manager RTMT can send email and page alerts when problems are detected. It connects directly to devices by using HTTP for troubleshooting system problems.

• The Cisco Conference Bridge application provides software conference bridge resources that can be used by Cisco Unified Communications Manager.

• The Cisco Unified IP Phone Address Book Synchronizer allows you to synchronize Microsoft Outlook or Outlook Express address books with Cisco Personal Address Book. After installing and configuring Cisco Personal Address Book, you can access this feature from the Cisco Unified IP Phone Configuration website.

• The Cisco Unified Communications Manager Locale Installer provides user and network locales for Cisco Unified Communications Manager, adding support for languages other than English. Locales allow you to view translated text, receive country-specific phone tones, and receive Tool for Auto-Registered Phone Support (TAPS) prompts in a chosen language when working with supported interfaces. This application is downloaded from the Cisco website as needed.

• The Cisco Dialed Number Analyzer is a serviceability tool that analyzes the dialing plan for specific numbers.

• Cisco Unified Communications Manager Assistant provides call-routing and display capabilities required by busy administrative assistants and their managers in a business environment. By combining a PC-based console application and various softkeys and display panes on Cisco Unified IP Phones, Cisco Unified Communications Manager Assistant can offer you job-specific tools to more efficiently manage calls in this environment. This function is also available as an XML service on the phone.

• The Cisco Unified Communications Manager JTAPI plug-in is installed on all computers that host applications that interact with Cisco Unified Communications Manager with the Java Telephony API (JTAPI). JTAPI reference documentation and sample code are included.

• Cisco Telephony Service Provider contains the Cisco Telephony Application Programming Interface (TAPI) service provider (TSP) and the Cisco WAV drivers that TAPI applications use to make and receive calls on the Cisco Unified Communications system.

SIP support is available in Cisco Unified Communications Manager with support for line-side devices, including IETF RFC 3261-compliant devices available from Cisco 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 7906G, 7911G,



7931G, 7941G, 7941G-GE, 7942G, 7945G, 7961G, 7961G-GE, 7962G, 7965G, 7970G, 7971G, and 7975G models, as well as the Cisco Unified IP Phone Expansion Module 7914. 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 Cisco Unity® and Cisco Unified MeetingPlace® solutions. CAC helps ensure that voice 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 allows remote device and system configuration. HTML-based online help is available for users and administrators. Cisco Unified Communications Manager 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. The Cisco Unified IP Phone 7931G initially supported in Cisco Unified Communications Manager 6.0 with Skinny Client Control Protocol (SCCP) is now optionally available with SIP. This phone provides functions that are commonly needed in the commercial and retail environments. It provides 24 lighted line keys and four interactive softkeys that guide you through call features and functions. In addition, it provides hard hold, redial, and transfer keys to facilitate simple and rapid call handling. SNMP is available to manage Cisco Unified Communications Manager, allowing managers to set and report traps on conditions that could affect service and send them to remote monitoring systems.

System Capabilities Summary

• Alternate automatic routing (AAR)

• Attenuation and gain adjustment per device (phone and gateway)

• Audio message-waiting indicator (AMWI)

• Automated bandwidth selection

• Autoroute selection (ARS)

• AVVID XML Layer (AXL) Simple Object Access Protocol (SOAP) application programming interface (API) with performance and real-time information

• Basic Rate Interface (BRI) endpoint support: Registers BRI endpoints as SCCP devices

• CAC: Inter- 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

• Call preservation - redundancy and automated failover - on call-processing failure

• Call recording

• 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, Global System for Mobile-Enhanced Full Rate (GSM-EFR), Global System for Mobile-Full Rate (GSM-FR) iLBC (internet Low Bitrate Codec), wideband audio (proprietary 16-bit resolution; 16-kHz sampled audio), and Advanced Audio CODEC (AAC) for use with Cisco TelePresence devices

• Digit analysis and call treatment (digit string insertion, deletion, stripping, dial access codes, digit string translation, and dial pattern *transformation)

• Database resiliency to increase feature availability for the following:

– Extension mobility

– Call forward all



– Message-waiting indicator (MWI)

– Privacy

– Device mobility

– Do not disturb

– End-user and Application User Certificate Authority Proxy Function (CAPF) for CTI

– Monitoring

– Hunt groups

• Device mobility changes in the location-specific information when a device moves within the cluster

• Dial-plan partitioning

• Distributed call processing

– Deployment of devices and applications across an IP network

– Virtual clusters of up to eight Cisco Unified Communications Manager servers for scalability, redundancy, and load balancing

– Maximum 7500 Cisco Unified IP Phones per Cisco Unified Communications Manager server and 30,000 per server cluster (configuration dependent)

– Maximum of 100,000 busy-hour call completions (BHCCs) per Cisco Unified Communications Manager 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

• Divert calls to voicemail (iDivert)

• 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, login, and logout

• Interface to H.323 gatekeeper for scalability, CAC, and redundancy

• IPv4

• 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 Communications Manager 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

• Off-premises extension (OPX)



• Outbound call blocking

• Out-of-band dual tone multifrequency (DTMF)

• Programmable line keys

• PSTN failover on route no availability [[is that ok? on route no availability? yes]]: AAR

• Q.Sig

– Alerting name specified in ISO 13868 as part of the Connected Name Identification Presentation (SS-CONP)

– Basic call

– ID services

– General function procedures

– Call back: ISO/IEC 13870: 2nd Edition, 2001-07 (completion of calls to busy subscriber [CCBS] and call completion on no reply [CCNR])

– Call diversion: 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 rerouting

– 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] and 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

– MWI

– Path replacement ISO/IEC 13863: 2nd Ed. 1998, and ISO/IEC 13974: 2nd Ed. 1999

• 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

– Secure conferencing is available to all members of the conference.

– Configurable operation modes: Nonsecure or secure modes can be configured.

– Device authentication: New model phones have an embedded X.509v3 certificate; a CAPF is used to install a 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 help ensure that they are not altered on the wire and can be trusted.

– Cisco Unified Communications Manager offers secure HTTP support for Cisco Unified Communications Manager Administration, Cisco Unified Communications Manager Serviceability, Cisco Unified Communications Manager User Pages, and Cisco Unified Communications Manager CDR Analysis and Reporting Tool.

– Privacy: Signaling and media are encrypted, including Cisco Unified IP Phone 7906G, 7911G, 7921G, 7940G, 7931G, 7941G, 7941G-GE, 7942G, 7945G, 7960G, 7961G, 7961G-GE, 7962G, 7965G, 7970G, 7971G, and 7975G models; Cisco Unified Survivable Remote Site Telephony; and MGCP gateways.

– Secure Sockets Layer (SSL) for directory: Supported applications include Cisco Unified Communications Manager BAT, Cisco Unified Communications Manager CDR Analysis and Reporting Tool, Cisco Unified Communications Manager Admin User Pages, Cisco Unified Communications Manager Assistant Admin Pages, Cisco Unified IP Phone Options Pages, Cisco Conference Connection, Cisco CTI Manager, Cisco Communications Manager Extension Mobility, and Cisco Communications Manager Assistant.



– A universal-serial-bus (USB) eToken containing a Cisco rooted X.509v3 certificate is used to generate a Certificate of 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 Communications Manager system administrator can disable HTTP and Telnet on IP phones.

– SIP trunk (RFC 3261) and line side (RFC 3261-based services)

– Cisco Unified SRST

• Shared resource and application management and configuration

– Transcoder resource

– Conference bridge resource

– Topological association of shared resource devices (conference bridge, music-on-hold [MoH] sources, and transcoders)

– Media termination point (MTP): Support for SIP trunk and RFC 2833

– Annunciator

• Silence suppression and voice activity detection (VAD)

• Silent monitoring

• Simplified North American Numbering Plan (NANP) and non-NANP support

• SIP trunk Call Admission Control (SIP CAC)

• T.38 fax support (H.323, MGCP, and SIP)

• Third-party applications support

– Broadcast paging: Through foreign exchange station (FXS)

– Simple Messaging Desktop Interface (SMDI) for MWI

– Hook-flash feature support on selected FXS gateways

– TSP 2.1

– 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

Note: Asterisks (*) in this list indicate SIP support for Cisco Unified Communications Manager 7.0. • *Abbreviated dial

• *Answer and answer release

• *Auto answer and intercom

• *Callback busy and no reply to station



• *Call connection

• *Call coverage

• *Call forward: All (off net and on net), busy, no answer, no bandwidth, and not registered

• *Call hold and retrieve

• Call join

• *Call park and pickup

• *Call pickup group: Universal

• *Call pickup notification (audible or visual)

• *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 chaining

• *Conference list and drop any party (impromptu conference)

• *Dialed-number display

• *Direct inward dialing (DID) and direct outward dialing (DOD)

• *Directed call park with BLF

• *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

• *Do not disturb (do not ring and call reject)

• *Drop last conference party (impromptu conferences)

• *Extension mobility support

• *Hands-free, full-duplex speakerphone

• *HTML help access from phone

• *Hold reversion

• *Immediate divert to voicemail

• *Intercom with whisper

• *Join across lines

• *Last-number redial (on and off net)

• *Log in and log out of hunt groups

• Malicious-call ID and trace

• *Manager-assistant service (Cisco Unified Communications Manager 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 Communications Manager 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 determine MWI status of manager phone



• *Manager-assistant service (Cisco Unified Communications Manager Assistant application) system capabilities: Multiple managers per assistant (up to 33 lines) and redundant service

• *Manager-assistant service now available on a Cisco Unified IP Phone with Cisco Unified Communications Manager 6.0

• *MWI (visual and audio)

• *Multiparty conference: Impromptu with add-on meet-me features

• *Multiple calls per line appearance

• *Multiple line appearances per phone

• *MoH

• *Mute capability from speakerphone and handset

• *On-hook dialing

• Operator attendant: Cisco Unified Communications Manager Attendant Console: Call queuing, broadcast hunting, and shared-line support

• *Original calling party information on transfer from voicemail

• *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 button barge

• *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

Cisco Unified Mobility

The Cisco Unified Mobility service helps mobile workers direct their inbound business calls to their IP phone number and initiate outbound business calls as if they were at their Cisco Unified IP phone - all from the mobile phone (or other remote phone destination). They can answer incoming calls on the desk phone or mobile phone, pick up calls between the desk phone and mobile phone without losing the connection, and originate enterprise calls from a mobile or other remote phone. Cisco Unified Mobility is included in Cisco Unified Communications Manager 6.0 and provides the following features: • Allowed and blocked call filters

• Caller identification

• Call screening and call divert

• Call tracing

• Cisco Mobile Voice Access

• Desktop pickup

• Directed call park through DTMF

• Mobile call pickup

• New mobility device model type

• Remote on and off control

• Reverse callback to non-mobile number



• Security and privacy for Cisco Unified Mobility calls

• Single enterprise voice mailbox

• Simultaneous desktop ringing

• System administrator-controllable user profile access

• Voice-based access with user identification and personal identification number protection

Summary of Administrative Features

• Application discovery and registration to SNMP manager

• AXL SOAP API with performance and real-time information

• Cisco Unified Communications Manager BAT (including new import and export capabilities)

• CDRs

• Cisco Unified Communications Manager CDR Analysis and Reporting Tool

• Call forward reason code delivery

• Centralized, replicated configuration database and distributed web-based management reports

• 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 voice-over-IP (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 CTI

• 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

• Ability to 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

• XML API for IP phones

• Zero-cost automated phone moves

• Zero-cost phone adds

• Data migration assistant

• Log partition monitor

• Disaster recovery framework

• Cisco Security Agent for Cisco Unified Communications Manager

• IP Security (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 Communications Manager BAT

• Scheduled trace collection

• User-defined events

• Real-time trace monitoring

• Enhanced upgrade process to minimize service downtime

• Enhanced installation process to minimize install time

• Installation answer file for no-touch installation

• Syslog to SNMP trap MIB

• Enhanced AXL SOAP API to modify the database

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 Communications Manager supports coexistence of SCCP and SIP phones, allowing migration to SIP while protecting investments in existing devices. Cisco Unified Communications Manager includes the following major SIP functions: • Native support of SIP devices

• CTI for Internet service provider (ISP) phones

• Presence information for SIP devices, including support for PUBLISH

• 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, 3323, 3325, 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, softphones, 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 Communications Manager to determine capacity.

• DLUs must be purchased to cover the number of devices connected to Cisco Unified Communications Manager.

• Third-party SIP devices require DLUs for operation with Cisco Unified Communications Manager.

Localization

The following user locales (languages) are supported: Arabic, Bulgarian, Catalan, Chinese (Hong Kong), Chinese (simplified), Chinese (traditional), Croatian, Czechoslovakian, Danish, Dutch, Estonian, Finnish, French, French (Canadian), German, Greek, Hebrew, Hungarian, Italian, Japanese, Korean, Latvian, Lithuanian, Norwegian, Polish, Portuguese, Portuguese (Brazilian), Romanian, Russian, Serbian , Slovak, Slovenian, Spanish, Spanish (Latin American), Swedish, Thai, and Turkish. 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.

Ordering Information

Software Upgrades

Cisco Unified Communications Manager 7.0 installation CDs and DVDs can be ordered for existing systems. Customers with a Cisco Unified Communications Software Subscription running Cisco Unified Communications Manager 4.1 to 6.1 who want to upgrade to Cisco Unified Communications Manager 7.0 can order upgrades using the Product Upgrade Tool located at http://www.cisco.com/upgrade. Customers planning an upgrade to Cisco Unified Communications Manager Version 6.0 should see the upgrade program for supported servers at:http://www.cisco.com/go/swonly. Hard disk capacity of 72 GB or greater and 2 GB of RAM are required.

Cisco Unified Workspace Licensing

This product is a part of Cisco Unified Workspace Licensing. Please visit http://www.cisco.com/go/workspace_licensing for more information.



Appendix B: Cisco Unity 7.0 for Microsoft Exchange

Product Overview - Cisco Unity® unified messaging is a foundational element in bringing unified communications solutions to enterprise-scale organizations. The solution provides anytime, anywhere collaboration through a broad range of productivity-enhancing features, a flexible platform including powerful migration tools for investment protection, and industry-leading capabilities for security and reliability such as Secure Messaging.


Anytime, Anywhere Collaboration

• Powerful unified or integrated messaging: Cisco Unity unified messaging integrates transparently with Microsoft Exchange, allowing you to handle all your messages - email, voice, and fax - through a single inbox using the Outlook email client. 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 also can reply to, forward, and save your messages - regardless of media type - in public or personal Microsoft Outlook folders with just a click of the mouse, decreasing response times and increasing organizational agility and customer service.

Integrated messaging allows you to access your voice messages through your Outlook email or any Internet Mail Access Protocol (IMAP) client, and does not require Active Directory or Exchange expertise. Integrated messaging users can still take advantage of the features inherent in the Cisco Unity system.

• Mobile access to voice messages: Cisco Unity unified messaging delivers all-in-one messaging for mobile users. Mobile workers using a Palm Treo or RIM BlackBerry device can simply double-click to play voice messages within their smartphone email applications. The Cisco Unity solution supports a variety of notification options that allow you to customize the way you are notified of new voice messages. Cisco Unity Unified Messaging for Microsoft Exchange users can access their voice messages using Cisco Unified Mobile Communicator, which integrates with Exchange to provide mobile access to messages. Even for users with basic mobile phones, the Cisco Unity solution is optimized to enhance mobile productivity. When you call in from a mobile phone, speech recognition allows for hands-free usage of the system. If a call is dropped because of a less-than-fully reliable mobile phone network, the Interrupted Session Recovery feature resumes, on the next call-in, the session where the call left off, reducing lost time.

Flexible Platform

• Migration at your own pace: Whether you need a rapid migration to IP telephony or require a more gradual pace, the Cisco Unity solution immediately improves productivity while allowing you to migrate at your own pace. Designed for an IP environment, this solution plays a central role in the migration of your telephony infrastructure from time-division multiplexing (TDM) to IP. The application interoperates with Cisco Unified Communications Manager and traditional telephony systems, including multiple-vendor private-branch-exchange (PBX) systems, at the same time to help you transition to IP telephony at your own pace and protect your existing infrastructure investments. In addition, Cisco Unity Session Initiation Protocol (SIP) integration provides native support for SIP proxy servers, designated SIP phones and clients, and SIP-enabled access gateways, to give SIP users access to the full array of benefits that the Cisco Unity application delivers.

• Networking capability: Cisco Unity unified messaging allows you to easily integrate your system with other voice messaging systems in your environment. The application includes a digital networking module that allows the system to connect to other Cisco Unity servers at the same site through the LAN, or remote sites using a WAN or the Internet. Digital networking makes communicating with co-workers at remote locations fast and efficient by allowing you to send subscriber-to-subscriber messages anywhere in the world. The Cisco Unity system supports Voice Profile for Internet Mail (VPIM [digital]) and Audio Messaging Interchange



Specification (AMIS [analog]) networking, which allow message interchange between disparate messaging systems that also support these industry-standard messaging protocols, helping to ensure a smooth system migration.

A powerful message networking option available with this solution is the Cisco Unity Bridge. With Cisco Unity Bridge, you can send subscriber-to-subscriber messages to anyone in your organization who resides on a TDM-based Avaya or Octel voicemail system supporting Octel Analog Networking. In addition, you can simply "reply to" a networked message with a single touch-tone key. With Cisco Unity Bridge, you can maintain advanced messaging capabilities on both systems as you migrate to the Cisco Unity system.

Secure and Reliable

• Secure Messaging: Cisco Unity unified messaging can encrypt messages as they are taken. You can then listen to the messages either through the telephone user interface (TUI) or from Microsoft Outlook, and the messages will be properly decrypted. With Secure Messaging, if a message is then forwarded outside the organization, the recipient of the forward will be unable to decrypt that message. Thus you can use Secure Messaging to prevent messages from leaving the organization.

Additionally, you can mark messages as private, in which case the encryption mechanism limits playback to only the original intended recipient. You can also configure encryption keys to expire after a set period of time, meaning messages will become unplayable records after the end of the expiry period even if copied to a computer hard drive. With Secure Messaging you can be assured that your security and compliance policies will be strictly adhered to while still allowing users the benefits of unified messaging.

• Resistance to Microsoft Exchange service interruptions: Cisco Unity unified messaging is designed to easily handle Microsoft Exchange service outages. The application uses the Cisco Unity Message Repository, which allows the system to continue taking new voice messages when the email system or network is offline. System subscribers also can retrieve these messages, minimizing service disruption. Additionally, because it keeps a local snapshot of the Active Directory environment, the Cisco Unity solution is not affected by Active Directory service concerns.

• Failover and standby redundancy: The Cisco Unity application supports configuration as a standby pair. In such a configuration - even in the case of a server failure - the system environment transparently fails over to the secondary Cisco Unity server, helping ensure high availability. Cisco Unity unified messaging can also support a capability called Standby Redundancy, which gives the system resilience during site-level failures. If a catastrophic site-level disaster occurs, you can manually switch the Cisco Unity system to a secondary site, allowing recovery of service within a short time window.

Features and Benefits of Cisco Unity 7.0

New for Cisco Unity 7.0

• Cisco Unity 7.0 is scalable to 200 ports and 15000 unified messaging users per server (depending on server type).

• If you log in from the same telephone number multiple times, the Cisco Unity system asks if you want the system to recognize you when dialing from the same number in the future.

• You can quickly modify your transfer settings through the voice user interface (VUI) or TUI so that calls are routed to you using predefined locations or calling party ID (CPID), also know as Follow Me.

• You can immediately reply back to external callers directly from the TUI based on CPID (Enhanced Caller Live Reply).

• You can strip forward introductions through the TUI prior to reforwarding a message.



• You can determine message durations before listening to the message.

• You can present most recently addressed subscribers first when addressing messages.

• When addressing a message, the Cisco Unity system returns the match name when dual-tone multifrequency (DTMF), also known as touch-tone, entry has reached uniqueness.

• You can include CPID and caller name in the subject line for outside caller messages (customizable subject lines).

• Newer Cisco Unified IP phones offer a constant message count.

• Outside callers can mark messages as private.

• Microsoft Exchange 2007 and Outlook 2007 IMAP are supported.

Interoperability and Availability

• VPIM support for multiple types of PBXs with one centralized Cisco Unity system provides for digital interoperability.

• AMIS support provides for analog interoperability.

• Cisco Unity Bridge offers interoperability with traditional Avaya or Octel voicemail systems.

• Network messages with Cisco Unity Express or Cisco Unity Connection with VPIM.

• Cisco Unity Message Repository manages new voice messages when the email system or network is offline.

• Q Interface Signaling Protocol (QSIG) and Digital Private Network Signaling System (DPNSS) support enhances integration with traditional PBXs.

• Failover capability prevents service disruption if the unified messaging server is unavailable, delivering enhanced reliability and serviceability.

Message Access from the TUI

• Both intuitive speech and "press or say" capabilities enable the use of speech commands to navigate menus and manage voicemail messages (Speech Access).

• You can automatically return to in-progress message composition or playback if you ended a session prematurely (Interrupted Session Recovery).

• You can screen voicemail messages as they are being recorded (Message Monitor).

• You can access your voicemail inbox through the IP phone interface. You can use the message locator to view the voicemail message queue or jump to a particular message in your inbox, view message header details, and play selected messages (Visual Voicemail for Cisco Unified IP Phones).

• You can play and process messages (repeat, reply, forward, delete, save, mark as new, hear day or time stamp, or skip to the next message).

• You can deliver messages to users at designated telephone numbers -- for example, home, cellular or mobile telephone, or remote-office telephone.

• You can reverse, pause, or fast forward during message playback.

• You can control volume and speed during message playback.

• You can pause or resume during message recording.



• You can address messages to multiple recipients.

• You can list all system subscribers in a central directory (Global Addressing).

• You can locate a message by number or name (Go to Message).

• You can record messages and specify them as regular, urgent, private, or future delivery.

• You can record messages and request a return receipt.

• You can switch between spelling the name and extension when addressing a message.

• You can immediately reply to messages by calling them back directly from the TUI (Live Reply).

• You can forward faxes to any fax machine from a touch-tone telephone.

Message Access from the PC

• You can access voice messages visually with IMAP client; note that you can deploy either Cisco Unity Inbox or IMAP client.

• With a digital video recorder (DVR)-style interface in email client, you can play, rewind, pause, or fast forward voice messages with a few mouse clicks.

• You can send voice and fax messages to anyone who can receive Internet email.

• You can download all message types and respond to or create new messages offline.

• You can save voice and fax messages along with email in public or personal Microsoft Exchange or Microsoft Outlook folders for a complete record of your communications.

• You can apply Microsoft Exchange Inbox Assistant rules to voice and fax mail.

End-User Features

• You can customize your message-notification options, manage personal greetings, or change passwords with Cisco Unity Assistant (the Cisco Unity Personal Communications Assistant web browser-based personal administrator).

• You can select conversation type; full or brief prompts are supported.

• You can change prompt and message playback speed.

• You can address and then record a message, or record and then address a message.

• You can record up to five personal greetings (alternate, busy, internal, off-hours, or standard).

• You can specify the order in which messages are presented over the phone, by message type (voice, fax, or email), urgency, or LIFO/FIFO.

• You can create private distribution lists and address messages to them through the TUI or by using Cisco Unity Assistant.

• You can set an expiration date for any personal greeting.

• You can manage an alternate greeting, require callers to listen to the full greeting, or notify users when a greeting is on.

• You can provide message notification for new messages through devices such as Simple Mail Transfer Protocol (SMTP) text, pagers, and phone destinations.



• You can provide message notification with Short Message Service (SMS) text messaging for mobile users; the Cisco Unity system supports the Short Message Peer-to-Peer (SMPP) 3.4 protocol for interoperability with all major SMS center providers.

• With a cascade message-notification feature, you can send additional notification types if a message is not retrieved.

• You can select whether or not message counts are announced; options include type, totals, saved, and new counts.

• You can specify whether or not the Cisco Unity system announces a transferred call.

• You can specify Call Forward to a personal greeting or busy greeting.

• You can specify an after-greeting action; after a subscriber greeting, callers can be directed to leave a message, sign in, or hang up, or they can be sent to call handlers, a directory handler, an interview handler, or a subscriber.

System Administration Overview

• You can integrate the Cisco Unity system with Cisco Unified Communications Manager and leading traditional telephone systems, even simultaneously, thereby paving the way for a smooth transition to IP telephony.

• The Cisco Unity system offers native support for SIP proxy servers, designated SIP phones and clients, and SIP-enabled access gateways.

• An intuitive browser-based system administration console and tools simplify installation, maintenance, and daily use and allow maintenance from any PC on the network.

• Cisco Unity Assistant (the Cisco Unity Personal Communications Assistant web browser-based personal administrator) allows IT staff to enable end users to manage more of their own accounts, saving time and decentralizing routine administration.

• Superior component-based server architecture provides a solid and flexible foundation for future growth.

• Innovative use of streaming media provides efficient audio delivery.

• Fault-tolerant system tools include robust security, file replication, event logging, and optional software Redundant Array of Independent Disks (RAID) levels 0-5.

• The Cisco Unity system offers full localization in U.S. English, French, German, and Japanese -- including system prompts, subscriber conversations, browser-based administration consoles, and product documentation.

• Localized telephone system prompts are available in multiple languages, including five dialects of English (Australian, Canadian, New Zealand, United Kingdom, and United States), Arabic (Formal), three dialects of Chinese (Cantonese, Mandarin Chinese [PRC], and Mandarin Chinese [ROC]), Czech, Danish, Dutch, Flemish, French (European and Canadian), German, Hungarian, Italian, Japanese, Korean, Norwegian, Polish, Portuguese (Brazilian and European), Russian, two dialects of Spanish (Latin American and European), and Swedish.

• The Cisco Unity system supports physical terminal line (tty) conversation for accessibility.

• The system is scalable to 200 ports and 15000 unified messaging users per server (depending on server type; refer to the Cisco Unity Supported Platforms list at: http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_data_sheets_list.html). These servers are then networked to support larger enterprise environments.

System Administration Features



• Alternate extensions are configurable by you or the system administrator.

• Alternate key mappings for message retrieval can help you transition from your existing voicemail system.

• Automatic gain control gives you consistent message volume playback levels.

• Billing ID is configurable on the system.

• You can browse to another Cisco Unity Administrator on a networked Cisco Unity server.

• The system supports call-holding queues.

• Call handlers can accept calls, play recorded prompts, route calls, and accept messages.

• Caller ID is supported.

• Call Routing and Automated Attendant features are configurable.

• Call Screening is configurable.

• Class of service support controls subscriber access to features.

• You can create subscribers individually or in bulk.

• Cross-Server Live Reply is supported for Cisco Unified Communications Manager deployments.

• Cross-Server Logon is supported for Cisco Unified Communications Manager deployments.

• Day and time stamps for messages are supported.

• Directory handlers can manage how callers search the directory.

• You can search the directory by spelling a subscriber name; entry of up to 24 letters is allowed.

• You can access your mailbox easily from your personal greetings by logging in to the TUI without entering your ID.

• Encrypted Skinny Client Control Protocol (SCCP) and Secure Real-Time Transport Protocol (SRTP) are supported for Cisco Unified Communications Manager integrations.

• Event logging is supported.

• Automatic and manual failover are supported.

• Automatic and manual failback are supported.

• Full mailbox warning is supported.

• Guided installation is supported.

• You can configure a list of observed holidays.

• Guest conversation is customized for the hotel and resort industry.

• A property management systems (PMS) interface is available (requires purchase of PMS hotel communication software from Percipia Networks).

• The Interview Handlers feature collects recorded input from callers.

• Identified subscriber messaging (ISM) between networked Cisco Unity servers in the same dialing domain is supported.



• You can configure how the Cisco Unity system handles messages that are interrupted by disconnected calls.

• A message-waiting indicator (MWI) is supported.

• You can move subscriber mailboxes without shutting down the Cisco Unity system.

• Multiple administrative levels are supported to control access to pages in the system administration GUI by class of service (read, modify, or delete rights).

• Multiple audio codecs are supported.

• Multiple time zones are supported.

• Music on hold is supported.

• Nondelivery or delivery receipt reason details are presented in the GUI inbox.

• OS, message store, and third-party software support includes:

• Message store support: Microsoft Exchange 2000, 2003, and 2007

• Windows 2000 Server on the Cisco Unity server

• Windows 2000 Advanced Server on the Cisco Unity server

• Windows Server 2003 on the Cisco Unity server (refer to release notes for details: http://www.cisco.com/en/US/products/sw/voicesw/ps2237/prod_release_notes_list.html)

• Windows Server 2003 Enterprise Edition on the Cisco Unity server (refer to release notes for details): http://www.cisco.com/en/US/products/sw/voicesw/ps2237/prod_release_notes_list.html)

• Support for data-protection software

• Support for remote-access software

• Support for monitoring software

• You can specify the public distribution lists to which new users will be added.

• Restriction tables are configurable.

• The Exclude Return Receipts Registry Controlled feature is supported.

• Schedules are configurable.

• You can self-enroll to set a password, record your voice name, and specify your directory listing.

• Subscriber licenses can be shared among networked Cisco Unity servers (license pooling).

• A status monitor gives the system administrators real-time status of fax and telephone ports, reports in progress, and system configuration.

• System broadcast messages are supported for single Cisco Unity server deployments and multiple server deployments.

• System greetings are configurable.

• Time stamps for 12- and 24-hour clocks are supported.

• The system automatically adjusts the time clock for daylight savings time.



• Cisco Unity unified messaging offers a TUI greetings administrator (Cisco Unity Greetings Administrator).

Fax

• Cisco Fax Server, an optional, full-featured fax solution based on the market-leading Captaris RightFax 9.3 product line, is available.

• For information about the supported third-party fax server hardware and software, visit: http://www.cisco.com/en/US/products/ps6178/index.html.

Security

• Secure Messaging enforces voicemail retention policies and prevents the compromise of voicemail messages with proprietary or confidential content forwarded to someone outside the enterprise.

• A host intrusion prevention system is supported; the Cisco Security Agent standalone agent protects Cisco Unity servers from worm and virus attacks. An optional Cisco Security Agent management console is also available.

• Password and personal-identification-number (PIN) security policy options enforce expiration, complexity, reuse, and lockout.

• An optional RSA Secure-ID 2-factor one-time PIN authentication server interface is available.

• Call-restriction tables prevent toll fraud.

• Security event logging and reports of failed login and account lockouts help detect "PIN cracker" attack attempts.

• SRTP and signaling encryption helps ensure secure communication between the Cisco Unity system and Cisco Unified Communications Manager.

• The Subscriber PIN Reset feature in Cisco Unity Assistant reduces help desk calls and operating expenses.

• Message archiving utilities enforce corporate electronic records-retention policies.

• Support for HTTPS provides secure web access to the Cisco Unity system.

Reports

• Administrative Access Activity Report

• AMIS Out Traffic Report

• AMIS In Traffic Report

• Call Handler Traffic Report

• Distribution Lists Report

• Event Log Report

• Failed Login Report

• Outcall Billing Report

• Port Usage Report



• Subscribers Report

• Subscriber Message Activity Report

• System Configuration Report

• Transfer Billing Report

• Unresolved References Report

• For a full list and description of Cisco Unity reports, refer to the Interface Reference Guide for the Cisco Unity Administrator Release 7.0 (With Microsoft Exchange), "Report Settings" chapter: http://www.cisco.com/en/US/docs/voice_ip_comm/unity/5x/interface_reference/guide/ex/50curg120e.html.

Licensing - All user and interoperability functions are now offered under a single, low-cost user license that can be used for either voicemail or unified messaging. Additionally, port and session capacity and failover redundancy licensing is available in two sizes: 32 ports or sessions and 200 ports or sessions.

System Requirements - Cisco Unity unified messaging runs on the Cisco media convergence servers, or their equivalents. Refer to the Cisco Unity Supported Platform List for hardware configuration and scalability requirements at:

http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_data_sheets_list.html.



Appendix C: Cisco IP Phones and Clients

Model

Cisco Unified IP Phone

3911G Cisco Unified IP

Phone 7906G 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 2-line x 24 char Pixel based 192 x 64 (monochrome)

Pixel based 192 x 64 (monochrome)

Pixel based 192 x 64 (monochrome)

Programmable (soft) keys

2 4 4 4

Fixed feature keys 8 2 2 2 Advanced features none none none none

Handsfree Yes 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 SIP SCCP SCCP SCCP Other Protocols supported

SIP SIP SIP

802.3af Yes Yes Yes No Codecs Supported G.711, G.729,

G.729a G.711a, G.711u, G.729a, G.729ab

G.711a, G.711u, G.729a, G.729ab

G.711a, G.711u, G.729a, G.729ab

DHCP Yes Yes Yes Yes 802.1p/q Yes Yes Yes Yes



Model Cisco Unified IP

Phone 7921 Cisco Unified IP

Phone 7925 Cisco 7937G Conference

Station

Integral switch n/a n/a No

Number of line keys 6 6 1

Display Digital, 16-bit graphical backlit

TFT Color, 2"

Digital, 16-bit graphical backlit

TFT Color, 2"

Pixel based – 255 x 128


2 2 3

Fixed feature keys 3 3 6 Advanced features WEP and IEEE

802.1X LEAP authentication, QoS,

USB support

WEP and IEEE 802.1X LEAP

authentication, QoS, USB support

Conference phone (Shown above with optional add-on

speakers)

Handsfree Yes - via earpiece Yes - via earpiece or bluetooth headset

Yes


Yes - visual display Yes - visual display No

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

Signaling Protocol SCCP SCCP SCCP Other Protocols supported

IEEE 802.11a/b/g IEEE 802.11a/b/g Bluetooth

802.3af Not applicable Not applicable Yes Codecs Supported G.711, G.729,

G.722, iLBC G.711, G.729, G.722, iLBC

G.711, G.729, G.722

DHCP Yes Yes Yes 802.1p/q Yes Yes Yes



Model

Cisco Unified IP

Phone 7940G

Cisco Unified IP Phone 7941G and 7941G-GE



Integral switch 10/100 10/100 (7941G);

10/100/1000 (7941G-GE)

10/100 10/100/1000

Number of line keys 2 2 lighted line-keys

2 lighted line-keys 2 lighted line-keys

Display Pixel based 145 X 80 (grey scale)

Pixel based 320 x 222 (grey scale)


Pixel based 320 x 240 (TFT color)


4 (+2 speed dial / line)




Fixed feature keys 8 8 8 9 Advanced features none higher-resolution,

more infrastructure integration options

higher-resolution, more infrastructure integration options


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 Codecs Supported G.711, G.729a G.711, G.729a G.711a, G.711u,

G.729a, G.729ab, G.722 and iLBC

G.711a, G.711u, G.729a, G.729ab, G.722 and iLBC




Model


7960G

Cisco Unified IP Phone 7961G and 7961G-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 6 lighted line-keys 6 lighted line-keys Display Pixel based 145

X 80 (grey scale)


320 x 222 (grey scale)

320 x 240, 12-bit color depth






Fixed feature keys 8 8 8 8 Advanced features additional line

keys with 7914 module




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 Codecs Supported G.711, G.729a G.711, G.729a G.711a, G.711u,


G.711a, G.711u, G.729a, G.729ab, G.722 and iLBC




Model Cisco Unified IP

Phone 7970G Cisco Unified IP Phone 7971G-GE


Integral switch 10/100 10/100/1000 10/100/1000

Number of line keys 8 lighted line-keys

8 lighted line-keys 8 lighted line-keys

Display 320 x 234, 12-bit color depth







Fixed feature keys 8 8 8 Advanced features color touch-

screen color touch-screen color touch-screen

Handsfree Yes Yes Yes Message waiting indication

Yes Yes Yes

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

SIP SIP SIP

802.3af Yes Yes Yes Codecs Supported G.711, G.729a G.711, G.729a G.711a, G.711u,


DHCP Yes Yes Yes 802.1p/q Yes Yes Yes



Model


Cisco Unified Video Advantage

Cisco IP Communicator


Expansion Module 7915

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 Codecs Supported DHCP Yes Yes Yes 802.1p/q Yes n/a n/a

Cisco Unified Communications System Proposal

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