Attachment B - Netech Corp (Cisco) RFP Response - Inter-University

Request for Proposal SSP_082311LLN

Next Generation IP-Based Voice Communications System (VoIP)

A Shared Services Project

Ohio University Bowling Green State University

Shawnee State University Ohio Board of Regents

OARnet

Request for Proposal #: SSP-082311LLN Due: 2:00PM EST, Tuesday, September 27, 2011

Laura L. Nowicki Issued by: OHIO University Procurement Services Athens, OH 45701 E-mail: [email protected] Date of Issue: Tuesday, August 23, 2011

2 Next Generation Voice | OHIOU-BGSU-SSU

Project Managers

Name University Phone Email

Michael Elliott Ohio University 740.593.2904 [email protected]

Michael Smith Bowling Green State University

419.372.9510 [email protected]

Jeff Blevings Shawnee State University 740.351.3275 [email protected]

Table of Contents

1 REQUEST FOR PROPOSAL ....................................................................................................................... 4

1.1 RFP OVERVIEW .......................................................................................................................... 4 1.2 OVERVIEW OF CURRENT ENVIRONMENT ......................................................................................... 6 1.3 REQUIREMENTS FOR THE VOICE COMMUNICATIONS SYSTEM .............................................................. 9

2 SYSTEM DESIGN OVERVIEW ................................................................................................................. 10

2.1 PROPOSED SYSTEM.................................................................................................................... 10 2.2 SYSTEM ARCHITECTURE .............................................................................................................. 13

3 IP COMMUNICATIONS SYSTEM SOFTWARE AND HARDWARE ........................................................... 33

3.1 SYSTEM SOFTWARE ................................................................................................................... 33 3.2 HARDWARE CONFIGURATION ...................................................................................................... 34 3.3 NETWORK INFRASTRUCTURE REQUIREMENTS................................................................................. 35 3.4 PSTN, LEGACY AND SIP INTEGRATION INTERFACES ......................................................................... 39 3.5 PROPOSED SYSTEM CABLING ....................................................................................................... 40 3.6 STATION HARDWARE.................................................................................................................. 41 3.7 SYSTEM RELIABILITY ................................................................................................................... 46 3.8 SYSTEM/STATION/USER FEATURES .............................................................................................. 47 3.9 DESKTOP CALL MANAGEMENT – VOIP SOFTPHONES ...................................................................... 48 3.10 E911 SERVICES ......................................................................................................................... 50 3.11 SYSTEM ADMINISTRATION .......................................................................................................... 51 3.12 SYSTEM MAINTENANCE AND UPGRADES ....................................................................................... 53 3.13 SYSTEM MONITORING AND DIAGNOSTICS ..................................................................................... 55

4 IMPLEMENTATION ............................................................................................................................... 56

4.1 PROJECT MANAGEMENT ............................................................................................................ 56


4.2 INSTALLATION REQUIREMENTS .................................................................................................... 58 4.3 FACILITY REQUIREMENTS ............................................................................................................ 60 4.4 TRAINING ................................................................................................................................. 61

5 VENDOR SERVICE .................................................................................................................................. 61

5.1 MAINTENANCE AND WARRANTY .................................................................................................. 61 5.2 LOGISTICAL SUPPORT ................................................................................................................. 62 5.3 REPAIR RESPONSE ..................................................................................................................... 62

6 CONFIGURATION/PRICING .................................................................................................................. 63

7 VENDOR REFERENCES .......................................................................................................................... 63

8 VENDOR PRESENTATIONS/PILOT/BAKE OFF ....................................................................................... 64

9 GENERAL .................................................................................................................................... 28

10 QUESTIONS REGARDING THIS PROPOSAL .................................................................................... 29

11 ADDITIONAL TERMS AND CONDITIONS ....................................................................................... 29

12 TIMELINE ................................................................................................................................... 32


1 REQUEST FOR PROPOSAL

1.1 RFP Overview

This Request for Proposal (RFP) document describes requirements of each university’s need for a native IP-based voice communications system. The proposed IP communications system must be able to support all the required call processing, voice messaging, unified communications, management and administrative features of this RFP. In addition, the proposed IP communication system must be capable of meeting anticipated growth without major system cost (i.e. forklift upgrade). This Request for Proposal is intended to provide a standard base from which to evaluate alternatives for communications systems and to allow the vendor flexibility in proposing the most appropriate and cost-effective system. The acceptance of this RFP does not obligate Ohio University, Bowling Green State University or Shawnee State University to purchase a system from any vendor. Each University may award separately and to different vendors. We are looking for a native IP based voice communication solution – respondents should use their knowledge and experience within the communications industry to recommend a creative solution that will meet or exceed each university’s requirements. Key System Requirements

IP-based Voice capabilities: Integration of voice applications within a converged Internet Protocol (IP) solution. Ability to provide highly reliable and available switching systems, a wide variety of interfaces to the PSTN (including SIP) and legacy TDM equipment, and choice of analog, digital or IP phones for endpoints including users, modems, fax machines, conference rooms, etc. Any proposed solution should allow a user to utilize multiple phones (physical set, softphone, mobile client, etc.).

Reliability: System must not have a single point of failure; allow outbound and inbound calls if the data network is down and better than five-9’s reliability. Vendor must supply phone sets with inline power (not local wall outlet) for power fail dial tone availability. Phones in remote locations must maintain all features in the event of WAN outage.

Voice Quality: Must be toll quality voice. Amount of latency to be expected must be documented. QoS must not require infrastructure upgrades. Please describe how QoS will be implemented, i.e. separate voice VLAN, etc.

Vendor Experience and Vision: Evaluation of the vendor's experience in building industry leading IP based voice systems.

Vendor support for Open System Standards: The solution should be committed to supporting open system industry standards, such as G.711, G.729, 802.1p and 802.1q, MGCP, RTP, TAPI, JTAPI, etc. IP handsets must use a standard signaling protocol. All features must be available on analog sets. System must support and be certifiable with any switch or router from any vendor.


Voice Messaging: Unified Messaging/Unified Communications integration should be included in your response. Each university’ messaging requirements may vary – please see section 1 for details.

Presence/IM: In conjunction with Unified Communications, the proposed solution should allow for all users to utilize a presence and instance messaging interface. Please describe how the proposed solution provides these features for all users and how the users would utilize.

Mobility: Integration of mobile devices (i.e. smart phones) is a requirement of the proposed solution. Please describe how your solution integrates Fixed Mobile Convergence (FMC) and wireless users (laptops/smart phones/ etc.). The ability for seamless connectivity between fixed and wireless telecommunications networks (cellular and university LAN/WAN) is key. These features should be compatible with industry leading smart phone devices such as RIM/Blackberry, Android and Apple IOS devices.

VoIP over Wireless (VoW, VoFI, wVoIP): The proposed system should include the ability for each university to perform voice calls (toll quality) via their respective 802.11 wireless network infrastructure. Please describe the needed infrastructure/requirements for this feature to be affective, include E9-1-1 location services.

Customer Service Call/Contact Centers: The proposed system should be enabled with powerful contact center features that will provide the universities with feature rich ACD queuing, web interface, IVR’s and instant messaging for callers.

Application Integration: Ability to integrate to CRM applications.

Fax to Email: The proposed system should integrate a fax to email feature (as part of Unified Communications). Please include all hardware and diagrams of the proposed integration.

Emergency - Event Notification: The proposed solution should allow the university to broadcast messages easily to all users of the system - simultaneously. These notices might be events or emergency situations.

Conference Bridge: The proposed solution should include conference call/bridging capabilities. The solution should be able to accommodate both ad hoc and scheduled conferences – with a web interface to manage the conferences. Please include specifics in terms of total allowed users (concurrent and conferences).

E9-1-1: Emergency response is very important to each university. Please see section 3 for each university environment. Please note that a detailed plan should accompany your RFP response in detailing E9-1-1 services which should include E9-1-1 services for both wired and wireless users.

System Administration: Single point of management from any point on the network for all components including the IP-PBX, voicemail, auto attendants, ACD and unified messaging system. Flexibility for efficient configuration changes to user profiles and IP telephone equipment through a standard browser-based interface.


Vendor Support/Service Capabilities: Remote serviceability, technical support of the entire telephone system and applications.

Scalability: Modular, cost-effective growth in end user devices (phones) and applications over the next 5-7 years. Fork-lift upgrade scenarios entirely should not be submitted as they may not acceptable in certain scenarios.

Security: Please provide detailed information regarding security of the proposed solution Please discuss security items such as: o OS type and version o Is the OS hardened? o How is the OS hardened? o Access rights o Password authentication o Security patches o Remote access o Voice/Data encryption o Virus protection o LAN/WAN security requirements (IPS, IDS, etc.)

Simplicity of Installation: Ease of installation and configuration will be important. Vendor should provide project management evidence of similar sized implementations.

Training and Usage: System must be easy to use, easy to learn and to administer.

Integration with OARnet Network To enable OARnet members to leverage their network investment in the OARnet backbone, the vendor proposed design should utilize the available OARnet bandwidth to the University to connect to the vendor provided VOIP gateway services. This approach should provide the following advantages: the university: 1) maximize their return on investment in their current bandwidth, 2) OARnet will be able to aggregate bandwidth from multiple university and deliver the connectivity to the Gateway service provider without the additional internet or network costs and 3)_OARnet can provision dedicated VLAN service for voice reducing possible contention with other IP services. The universities participating in this RFP reserve the right to accept or reject this portion of the response, however Respondents are encouraged to include this option to take advantages of the cost benefits associated with this approach. Vendors should provide any additional costs they may incur to connect to the OARnet network

1.2 Overview of Current Environment

1.2.1 Locations and Users

This section describes Ohio University’s current environment including the name and number of locations and the number of users at each. Values may increase or decrease after this RFP is written

Ohio University


Site # Site Address Type of PBX Analog User (A) Digital Users (D)

Voicemail users

1 OHIO UNIVERSITY ATHENS Athens Ohio

Ericsson Aastra MD110

A – 10,000 D – 4,000

4,500

2 OHIO UNIVERSITY LANCASTER

Lancaster Ohio

Nortel CS1000 A – 300 D – 300

<200

3 OHIO UNIVERSITY PICKERINGTON

Pickerington Ohio

Extension of Lancaster

A – 75 D – 75

<100

4 OHIO UNIVERSITY CHILLICOTHE

Chillicothe Ohio

NEC NEAX 2000 IPS

A – 300 D – 300

<200

5 OHIO UNIVERSITY ZANESVILLE

Zanesville Ohio

Intertel Axxess A – 300 D – 300

<200

6 OHIO UNIVERSITY EASTERN St. Clairsville Ohio

ComDial Vertical 15C

A – 300 D – 300

<200

7 OHIO UNIVERSITY SOUTHERN

Ironton Ohio

Nortel Option 11c

A – 300 D – 300

<200

8 OHIO UNIVERSITY PROCTORVILLE

Proctorville Ohio

Extension of Southern

A – 75 D – 75

<100

Bowling Green State University

Site # Site Address Type of PBX Analog User (A) Digital Users (D) VoIP (IP) Virtual (V)

Voicemail users

1 Bowling Green State University

Bowling Green Ohio

Avaya S8700 CM 2.2

A – 2000 D – 3200 I – 300 V – 200

4000

2 Firelands Campus Huron Ohio Survivable Remote of main PBX

D – 200 A - 150

0

3 Levis Commons Campus Perrysburg Ohio

G300 Remote gateway

A – 16 D – 24

0

Shawnee State University

Site # Site Address Type of PBX Analog User (A) Digital Users (D) VoIP (IP)

Voicemail users

1 Shawnee State University Portsmouth, Ohio

Nortel 61C Meridian Mail

A – 683 D – 457

470

1 Shawnee State University Portsmouth, Ohio

Cisco CM 6.1 Unity 5.0

A - 126 D - 0 IP - 34

29


1.2.2 Trunking Information

This section describes current trunking including the types and number of PRI trunks at each location. Please note that this may not be an exhaustive list of all trunks in the telephony realm at Ohio University. Values may increase or decrease after this RFP is written.

Ohio University

Location Type of Trunk # of Circuits

OHIO UNIVERSITY ATHENS Local PRI 13 PRI’s

Long Distance PRI 6 PRI’s

OHIO UNIVERSITY LANCASTER

Local / Long Distance PRI 1 PRI

OHIO UNIVERSITY PICKERINGTON

Local / Long Distance PRI 6 Ground Start Trunks

OHIO UNIVERSITY CHILLICOTHE


OHIO UNIVERSITY ZANESVILLE Local / Long Distance PRI 3 PRI’s

Long Distance PRI

OHIO UNIVERSITY EASTERN

Local PRI 1 PRI (inbound), Analog trunks for outbound

OHIO UNIVERSITY SOUTHERN


OHIO UNIVERSITY PROCTORVILLE

Local / Long Distance 4 Analog trunks 2 Ground Start Trunks




Local PRI 9 PRI’s


Long Distance PRI 4 PRI’s

Firelands Campus Local PRI 1 PRI

Levis Commons Campus Local 4 Analog lines

Shawnee State University


Shawnee State University Local PRI Nortel Switch 2 PRI’s

Long Distance PRI Nortel 1 PRI


Switch (Quest)

Local PRI Cisco Switch 1 PRI

1.2.3 Voicemail and Unified Messaging

Ohio University Ohio University currently has an AVST CallXpress 8x platform. Please describe integration that Ohio can take advantage of while continuing to use AVST for their voicemail needs. Bowling Green State University Bowling Green State University currently has Avaya Intuity Audix for voicemail. This need not be used in future systems, but may be for a phased implementation. Shawnee State University Shawnee State University currently has Meridian Mail supporting the Nortel Switch users and Cisco Unity providing support for the Cisco Call Manager users. We would like a system that would integrate both systems and would eventually be the main system for both systems.

1.3 Requirements for the Voice Communications System

Ohio University Ohio University seeks a VoIP solution that will be able to provide voice/call control with feature rich collaboration tools. All existing telephones should be replaced with either equivalent new analog or IP phones that support all basic and advanced telephony features. VoIP softphones and Unified Communications are expected to be part of any proposed system. Ohio University’s remote locations should be connected to the main location via IP with current DS-3 or T1 circuits. Each location should be able to access all the features and functionality available at the main site. System directories, class of service for telephony capabilities, trunk group access, should apply to all locations. Bowling Green State University Bowling Green State University requires a communications system that will, at a minimum, be able to provide voice and messaging services to all staff and associated personnel. This should be accomplished with a mix of VOIP telephone stations, analog dial tone, and VOIP softphones. Unified messaging is also required. Bowling Green State University’s remote locations are to be connected in such a manner as they are a part of the system that only require separate management for local survivable purposes. All phone numbers should be centrally managed and seen by the system the same, regardless of the location. Connections from the main campus system(s) to the various remote locations should utilize existing IP data networks. Shawnee State University


Shawnee State University seeks a solution that integrates its current communications systems with a Voice over Internet Protocol (VoIP) system that will eventually become the communications system for Shawnee State University. We would like to see a phased in approach with the phone set as the departments purchase new sets they would move to the VoIP platform and phones.

2 SYSTEM DESIGN OVERVIEW

As a shared services project of the IUC (Inter University Council) the proposed solution should be designed with the expectation of each university may be completely segregated from the others, however you may respond as a hosted solution (partially or full) or an anchor-tenant environment, whereas Ohio University may provide telephony services to the other universities, or vise-a-versa.

2.1 Proposed System

2.1.1 Provide a brief description of the proposed system. Please include diagrams. If you are proposing an anchor-tenant or hosted solution please detail (along with diagram) how this would work and how each university can take advantage of the services in such an environment.

Response: The Cisco Unified Communications Solution is a comprehensive, secure IP

communications solution that unifies voice, video, data, and mobile applications on fixed and

mobile networks, delivering an easy-to-use, media rich collaboration experience across the

University’s workspaces. These applications use the network as a platform to enable users to

collaborate every time, everywhere, while helping to reduce costs.

With Cisco Unified Communications, the University can:

Provide productive, collaborative, and highly mobile communications anytime,

anywhere, on a variety of devices, both wired and wireless.

Seamlessly pass calls between devices, inside and outside the office.

Save time with a single voicemail inbox (including calls sent to mobile phones).

Use the network as a secure platform for all communications (voice, video, data, and

mobility) so that you can extend your reach and do more with existing resources while

reducing the cost of maintaining aging disparate legacy systems in your facilities.

Enable faculty and staff to interact more effectively and access the services they need,

when they need them.

Connect the University’s end users with important resources, services, and information to

foster economic development.

Offer an extensive set of additional applications to smoothly integrate inbound and

outbound voice applications with Internet applications (real-time chat, Web

collaboration, and email) that will allow the University to continue to enhance

performance and productivity into the 21st Century.


Cisco Unified Communications has been documented and proven to offer several benefits to

educational customers of all sizes and types. We are re-defining communications by leveraging

and continuing to enhance the power and intelligence of the network. Our innovations help

create a high-performance system with streamlined processes, increased productivity, and a true

competitive advantage for our customers.

The Cisco Unified Communications solution can help the University more efficiently and

effectively serve students and faculty in many ways including:

Administrative Efficiency: Cisco Unified Communications provides for streamlined

operations and processes, as well as for increased productivity.

Safety and Security: Cisco Unified Communications provides the ability to monitor and

protect staff, students, and property; to immediately respond to security incidents; and the

capability for mass internal and external notifications.

Improve accessibility to information and experts: Cisco Unified Communications

supports a variety of features to improve collaboration and information transfer. These

include presence technology that enables employees to quickly determine the best

method for reaching coworkers; single number reach; unified messaging; sharing of

presence information with third-party applications; and the ability to transfer and route

calls easily.

Reduce costs: Cisco Unified Communications can both improve employee productivity

and provide direct cost savings. Sources of cost savings include reduced IT network

management requirements, elimination of multiple leased lines throughout school

facilities, and the ability to shift from third-party conferencing services to internal

conferencing. With flexible communications, you can also improve the productivity of

remote and mobile employees and reduce recurring travel expenses.

Increase productivity and speed up task completion and issue resolution: Through

customizable widgets, thin clients, Web clients, and a variety of devices and applications,

Cisco Unified Communications delivers a consistent, unified experience across a range of

devices and environments, streamlining and improving employee communications

wherever employees work.

Protect investments: With its open, interoperable framework, Cisco Unified

Communications provides investment protection and simplifies process integration. For

example, Cisco Unified Application Environment allows customers and partners to

rapidly develop new collaborative applications independent of language, platform, or

transport.

Cisco Unified Communications offers multiple deployment options that interoperate with what

you have today, let you migrate at a pace that’s right for your facilities, and maximize your

investment for continued growth and success.

2.2 Cisco Unified Communications Manager

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 that provides traditional telephony features as well as


advanced capabilities, such as mobility, presence, preference, and rich conferencing services.

This powerful call processing solution can help:

Simplify your voice systems by replacing old Private Branch Exchange (PBX) and key

systems with unified communications; you can cut costs and dramatically streamline

provisioning and maintenance

Build productivity with feature-rich unified communications that help workers spend less

time chasing people, and more time being productive

Enable mobility with software that has embedded unified mobility capabilities so mobile

workers can remain productive wherever they are

Improve collaboration with a click; start an IM session, initiate a phone call, and establish

a videoconferencing call more easily.

Cisco Unified Communications Manager creates a unified workspace that supports a full range

of communications features and applications with a solution that is highly:

Scalable: Each Cisco Unified Communications Manager cluster can support up to 30,000

users

Distributable: For scalability, redundancy, and load balancing

Available: Support business continuity and improve collaboration with high availability

that provides a foundation for multiple levels of server redundancy and survivability.

As your needs evolve, Cisco Unified Communications Manager continues to evolve to meet

those needs. Cisco Unified Communications Manager Version 8.0 aims to lower the total cost of

ownership for organizations and improve the communications experience for end users as well as

system administrators.

Central Design Primary Data Center

The primary data center will house servers for all IP Telephony applications as well as voice

gateways to meet PSTN trunking requirements. In the event of a failure of the primary location,

clustered call control and voicemail will provide full functionality to <CUSTOMER

ACRONYM>. Audio bridging is designed redundantly to address <CUSTOMER

ACRONYM>’s requirements leveraging two distinct conferencing resource pools. If one system

fails, the other can meet the full capacity requirements as required. Cisco Emergency Responder

servers provide consistent location information for emergency calls across a shared database.

Secondary Data Center

The servers located in the secondary data center are clustered together with servers in the

primary data center to provide load balancing, redundancy, and fault tolerance. Clustered servers

can have separate Layer 3 addresses, providing design and routing flexibility.

Site Design: A per-location migration strategy allows for a simplified deployment model. Based

on the information provided by <CUSTOMER ACRONYM>, integration requirements with

existing systems will be minimized as each location will receive new equipment. Integration

requirements may be met by QSIG trunks to existing PBX systems. Cisco ISR G2 routers at each

branch location provide improved (50 Mbps minimum) sustained WAN performance, scalability,

PSTN termination, SRST, security, and routing on a single converged platform. New Cisco PoE

switches will be installed. These high-performance switches automatically detect Cisco phones


and provide the exact amount of power required for each phone model. This eliminates the need

to run additional electrical outlets to each phone location. IP addresses, DHCP scopes, VLANs

(where necessary dot1q trunking), and subnets will be added to the existing network

routing/switching to accommodate telephony devices. QoS will be applied end-to-end to ensure

voice and video quality is protected.

All sites that do not have a local Unified Communications Manager server onsite, in the event of

a WAN outage will place, receive calls, and access voicemail via Survivable Remote Site

Telephony, a service provided by the local router. In the event of the failure of a Unified

Communications Manager, the phones will merely use the redundant server automatically.

2.3 System Architecture

2.3.1 Provide a brief description and discussion of your system architecture. Describe your philosophy on open architecture and your ability to support other vendors’ equipment. Provide a diagram of the system architecture.

Response: The following table lists protocols, open/industry standard, and proprietary, supported

by the proposed solution, indicating which key components these apply to and that to which it

interfaces.

Interfaces to other Voice, Video and Data Solutions

Component Interface Protocol

WAN

connectivity

Bandwidth control RSVP Agent, IEEE 802.1q/p, DiffServ

LAN

connectivity

Voice and video quality, Power

over Ethernet support

Cisco Discovery Protocol, IEEE 802.3af,

Cisco inline power, DiffServ, IEEE 802.1q/p

Security Authentication and transport

integrity

SRTP, TLS. SSL X509 v3, HTTPS, IPSec

Call-control

software

Network interoperability DPNSS, QSig, SIP, E1, T1, PRI, BRI.

Call Control Call control and CTI SCCP, H.323, SIP, MGCP, JTAPI, TAPI,

XML, VXML,

Endpoint CTI SCCP, HTTP, TAPI, XML, Java

Video and TV Video conferencing H323, H.320, H.261, H.263, T.120

Mobility 802.11b/g

Messaging Messaging interoperability SIP, VPIM, AMIS, Cisco Unity Bridge,

Octelnet, IMAP, T.37 fax, T.38 fax,

Collaboration Data collaboration, instant

messaging, conferencing

SIP, SIP/SIMPLE, CSTA, T.120

Contact Center CTI

Network

Management

Collection, management and

publishing of data

SNMP, HTTP, HTTPS, SOAP, XML


2.3.2 Describe how the system integrates with voice services with the converged Internet Protocol network including the use of standards and the support for analog and IP endpoints for users, modems, fax machines, etc.

Response: The Cisco-proposed solution supports several open-packet telephony standards that

allow customers, partners, and developers to easily extend the features and capabilities of their

communications environments. Most applications are supported using a combination of inherent

IP phone services, Extensible Markup Language (XML), Telephony Application Programming

Interface (TAPI), and Java Telephony API (JTAPI) standards.

The proposed Cisco VG224 Analog Phone Gateway combines a high-density RJ21 analog

interface with Cisco IOS Software manageability to deliver a cost-effective platform that

maximizes the functionality of existing analog phone equipment in a Cisco Unified

Communications system deployment.

Cisco Unity Connection can interact with a fax server directly through Simple Mail Transport

Protocol (SMTP). Inbound faxes are received by the fax ser and routed to the Unity Connection

server through SMTP. Similarly, faxes are routed to a fax server through SMTP for rendering

and outbound faxing.


2.3.3 Describe how the delivers reliability for voice services including maintaining dial tone during WAN outages, failure of the systems Windows based servers, and power outages.

Response: Survivable Remote Site Telephony (SRST) provides high availability for voice

services by providing a subset of the call processing capabilities within the remote office router

and enhancing the IP phones with the ability to “re-home” to the call processing functions in the

local router if a WAN failure is detected. Under normal operations, the remote site connects to

the central site via an IP WAN, which carries data traffic, voice traffic, and call signaling. The IP

phones at the remote site exchange call signaling information with the Unified CM cluster at the

central site and place their calls across the IP WAN. The branch router or gateway forwards both

types of traffic (call signaling and voice) transparently and has no knowledge of the IP phones.

If the WAN link to the branch office fails, power failure, server failure, or if some other event causes

loss of connectivity to the Unified CM cluster, the branch IP phones re-register with the branch router

in SRST mode. The branch router queries the IP phones for their configuration and uses this

information to build its own configuration automatically. The branch IP phones can then make and

receive calls either internally or through the PSTN. The phone displays the message “Unified CM

fallback mode,” and some advanced Unified CM features are unavailable and are grayed out on the

phone display.

When WAN connectivity to the central site is re-established, the branch IP phones automatically

re-register with the Unified CM cluster and resume normal operation. The branch SRST router

deletes its information about the IP phones and reverts to its standard routing or gateway

configuration.

Under normal operations shown in the left part of the diagram below, the branch office connects

to the central site via an IP WAN, which carries data traffic, voice traffic, and call signaling. The

IP phones at the branch office exchange call signaling information with the Unified CM cluster

at the central site and place their calls across the IP WAN. The branch router or gateway

forwards both types of traffic (call signaling and voice) transparently and has no knowledge of

the IP phones.


Survivable Remote Site Telephony or Unified CME in SRST Mode

If the WAN link to the branch office fails, or if some other event causes loss of connectivity to

the Unified CM cluster, the branch IP phones re-register with the branch router in SRST mode.

The branch router, SRST, or Unified CME running in SRST mode, queries the IP phones for

their configuration and uses this information to build its own configuration automatically. The

branch IP phones can then make and receive calls either internally or through the PSTN. The

phone displays the message “Unified CM fallback mode,” and some advanced Unified CM

features are unavailable and are grayed out on the phone display.


For additional information and a listing of features supported in SRST mode, please visit:

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps2169/data_sheet_c78-

520521.html.

Local Failover Deployment Model

The local failover deployment model provides the most resilience for clustering over the WAN.

Each of the sites in this model contains at least one primary Unified Communications Manager

subscriber and one backup subscriber. The maximum number of phones and other devices will

be dependent on the quantity and type of servers deployed. The maximum total number of IP

phones for all sites is 30,000.

Example of Local Failover Model

Observe the following guidelines when implementing the local failover model:

Configure each site to contain at least one primary Unified Communications Manager

subscriber and one backup subscriber.

Configure Unified Communications Manager groups and device pools to allow devices

within the site to register with only the servers at that site under all conditions.

Under a WAN failure condition, sites without access to the publisher database will lose

some functionality. For example, system administration at the remote site will not be able

to add, modify, or delete any part of the configuration. However, users can continue to

access the user-facing features listed in the section on Unified CM Publisher.

Under WAN failure conditions, calls made to phone numbers that are not currently

communicating with the subscriber placing the call will result in either a fast-busy tone or

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps2169/data_sheet_c78-520521.html



a call forward (possibly to voicemail or to a destination configured under Call Forward

Unregistered).

The maximum allowed Round-Trip Time (RTT) between any two servers in the Unified

Communications Manager cluster is 80 msec.

Note: At a higher round-trip delay time and higher Busy Hour Call Attempts (BHCA),

voice cut-through delay might be higher, causing initial voice clipping when a voice call

is established.

A minimum of 1.544 Mbps (T1) bandwidth is required for Intra-Cluster Communication

Signaling (ICCS) for 10,000 BHCA between sites that are clustered over the WAN. This

is a minimum bandwidth requirement for call control traffic, and it applies to

deployments where directory numbers are not shared between sites that are clustered over

the WAN. The following equation may be used as a guideline to calculate the bandwidth

for more than 10,000 BHCA between non-shared directory numbers at a specific delay:

RTT delay in msec.

This call control traffic is classified as priority traffic. Priority ICCS traffic is marked

with IP Precedence 3 (DSCP 24 or PHB CS3).

In addition to the bandwidth required for ICCS traffic, a minimum of 1.544 Mbps

(T1) bandwidth is required for database and other inter-server traffic for every

subscriber server remote to the publisher.

For customers who also want to deploy CTI Manager over the WAN, the following

formula can be used to calculate the CTI bandwidth (Mbps):

Voicemail High Availability

The Cisco Unity Connection cluster feature provides high availability voice messaging through

two Connection servers that are configured in a cluster. Under normal conditions, the Connection

servers are both active so that:

The cluster can be assigned a DNS name that is shared by the Connection servers.

Clients such as email applications and the Web tools available through the Cisco Personal

Communications Assistant (PCA) can connect to either Connection server.

Phone systems can send calls to either Connection server.

Incoming phone traffic load is balanced between the Connection servers by the phone

system, PIMG/TIMG units, or other gateways that are required for the phone system

integration.

PSTN High Availability

Outbound dialing is protected <CUSTOMER ACRONYM> -wide via Route Lists within

Unified Communications Manager. Route Lists allow calls to leverage a pool of

gateways in a prioritized list. If one gateway is unavailable or oversubscribed, the call is

placed using another within the previously configured list. This permits <CUSTOMER

ACRONYM>to provide redundant pools of PSTN resources to callers, utilizing resources

anywhere across <CUSTOMER LOCATION>.


For inbound calls, in the event of a failure affecting a trunk where phone numbers are

specifically associated with a unique connection, failover mechanisms must be

established in cooperation with the local PSTN provider.

2.3.4 Describe the vendor’s experience in building and delivering Voice over IP (VoIP) solutions.

RESPONSE: Cisco Systems is the world leader in IP-based voice communications. Starting

with an install base of zero in 1999, Cisco has become the worldwide leader in voice

communications market share. <PARTNER ACRO> finds this success particularly relevant

for our clients, given that Cisco sells only IP-based communication systems. Yet, compared

with the traditional players in the voice market selling both IP and TDM solutions, Cisco is

the leader in worldwide shipments.

The most interesting point to consider in this market transition is that 12 years ago there was

no one clamoring to converge voice, video, and data networks into a single application

infrastructure. The established players in the voice market were serving their customers well

and generally their customers were happy with the traditional TDM-based offerings they had

been deploying for the past 25 years. It speaks to the value proposition of the Cisco offering

that in this relatively short period of time, Cisco has shipped over 40 million phones to over

100,000 customers worldwide. Cisco is currently displacing over 35,000 TDM phones every

business day. The move to IP is now an everyday event, and Cisco has led the charge. Only

a truly actionable business case and a compelling vision for future applications could

completely disrupt a seemingly placid and stable marketplace. However, Cisco’s statistical

evidence shows that this market transition is not just underway, but is now mainstream as

customer acceptance continues to accelerate and traditional competitors scramble to convert

their offerings to IP. As we continually scan the market for the best offerings for our clients,

we have not found a solution that meets the breadth of the Cisco offering in terms of

applications, availability, security, and scalability.

Cisco is the market leader in IP communications and offers a unique, end-to-end approach that

set us apart from the competition. Cisco offers the following strengths over competing solutions:

Industry-leading experience: As of September, 2011, Cisco has more than 100,000

customers using our Unified Collaboration platform. We offer a proven technical model

that is widely deployed within our own network (which itself supports 55,000 unified

communications phones and video devices). We sustain our solution by acquiring new

technologies as they emerge and investing more heavily than competitors in annual

research and development. The following statistics reflect the proven track record and

broad range of Cisco Unified Collaboration capabilities:

o 400+ customers have deployed more than 5,000 IP phones

o 22M+ Unified Messaging seats

o 2M+ Contact Center agents


o 470K+ MeetingPlace licenses

o 1.2+ Billion meeting minutes per month

o 1M+ Cisco WebEx mobile app downloads

End-to-end solution: Cisco Unified Collaboration is an end-to-end solution that includes

network equipment, telephony systems, IP phones, and associated applications. Studies

show that implementing a network with a single, primary vendor helps customers achieve

a better Return on Investment (RoI) than they would with a multivendor solution.

Industry-leading security: Cisco is the only vendor that provides systemic, end-to-end

security. This comprehensive security starts in the network itself and extends all the way

to call control, endpoints, and applications using industry-standard technologies. Cisco’s

integrated systems approach to security has been demonstrated in independent tests to be

the most secure solution available today.

Highly interoperable solutions: Cisco supports open, standards-based network solutions

to deliver industry-leading interoperability, which protects and extends your investment.

In addition, Cisco is committed to integrating Cisco Unified Communications software

with a variety of applications and products from other vendors, including Microsoft,

IBM, and Oracle.

First-class partnerships: Cisco maintains partnerships with industry IT leaders to enable

educational institutions to deploy a highly adaptable network infrastructure and

innovative applications that help you get the most value from your infrastructure

investments.

2.3.5 Describe the systems support for open standards including support for open standards for integration with existing voice equipment.

Response: The following table identifies the standards which the Cisco solution supports.

Cisco Support for Industry Standards

Recommendation Status

G.711 Compliant

G.722 Compliant

G.723 Compliant

G.726 Compliant

G.728 Compliant

G.729 Compliant

G.729a Compliant

H.323 V4 Compliant

T.120 Compliant

Q.931 Compliant

802.1d Compliant

802.1p Compliant

802.1q Compliant

802.3 Compliant


Recommendation Status

SNMP Compliant

FAX - Group 3 Compliant

FAX - Group 4 Compliant

T.37 Compliant

T.38 Compliant

IP Precedence Compliant

Differentiated Services Compliant

RSVP Compliant

Weighted Fair Queuing Compliant

CBWFQ Compliant

PQWFQ Compliant

RED Compliant

Weighted RED Compliant

RTP Compliant

CRTP Compliant

RTCP Compliant

RTSP Compliant

Policy-Based Routing Compliant

Committed Access Rate Compliant

IPv6 Compliant

MGCP Compliant

H.225 Compliant

H.245 Compliant

TCP/IP Compliant

UDP/IP Compliant

DHCP Compliant

DCL Compliant

DNS Compliant

2.3.6 Describe the architecture of the proposed voicemail solution including how voice mail is accessed by users from their extension, remotely, and from their desktop computer. For Ohio University, a response is only needed if there is no integration with their AVST solution.

Response: Cisco Unity® Connection is a feature-rich voice and unified messaging platform based on the same Linux Unified

Communications Operating System as Cisco Unified Communications Manager. With Cisco Unity Connection, you can access and manage voice messages in a variety of ways, using your email inbox, web browser, Cisco Jabber®, Cisco Unified IP Phone, Cisco Cius™ business tablet, smartphone, Cisco Unified Personal Communicator, and more. Cisco Unity Connection also provides robust speech-recognition features for when you are mobile, so you can manage your voice messages hands- and eyes free.

Features and Benefits

Powerful Unified Messaging

At its core, Cisco Unity Connection is a powerful unified messaging system with many advanced capabilities that you can customize to maximize your individual and team productivity. You can personalize communications options and interact with the system to manage calls and messages in the way that is most comfortable and convenient for you. The flexible user interface makes messaging more efficient for "power users" and occasional voicemail users alike. For example, you can even customize your telephone user interface (TUI) and touch-tone mappings to make migration from traditional voicemail systems much easier.

Speech-Enabled Messaging


To maximize the productivity of mobile workers, Cisco Unity Connection offers a natural and robust speech-activated user interface that allows you to browse and manage your voice messages using simple, natural speech commands. In addition, the Speech Connect for Cisco Unity Connection feature is a built-in speech-enabled Automated Attendant that enables you to call other Cisco Unity Connection users or personal contacts by simply using your voice. The Cisco SpeechView feature of Cisco Unity Connection converts voice messages to text and delivers the text version of the voice message to your email inbox, allowing you to read your voice messages and take immediate action.

Powerful Desktop Message Access

Manage your voice messages from a variety of devices and locations, whichever best suits the way you work: • Unified messaging

– Voice messages in Cisco Unity Connection and Microsoft Exchange mailboxes are synchronized; they are available for Microsoft Exchange 2003, 2007, and 2010, the Microsoft Business Productivity Online Suite (Dedicated (BPOS-D) environments), and Microsoft Office 365.

– You can play, delete, record, reply to, and forward voice messages from within Microsoft Outlook using the ViewMail for Outlook plug-in that is included with Cisco Unity Connection.

– You have text-to-speech (TTS) access to Exchange email.

– You can access your Exchange calendars and contacts.

– The message-waiting indicator (MWI) and heard/unheard message status are synchronized.

– Secure, private messages are supported on the phone and Microsoft Outlook email folder.

– You can access voice messages in your mobile email inbox, such as on BlackBerry smartphones.

• Desktop and mobile collaboration application voicemail integration

– You can access your voice messages directly from the Cisco Jabber and Cisco Unified Personal Communicator client.

– You can access your voice messages from IBM Lotus Sametime using the Cisco Unity Connection for IBM Lotus Sametime plug-in.

– You can access your voice messages from Cisco WebEx® Connect using the Cisco Unified Communications Integration for WebEx Connect application.

– You can use the integrated media player to play and delete messages.

– You can easily access presence and availability information about the person you are calling in the Cisco Unified Personal Communicator, Cisco Jabber, IBM Lotus Sametime, and Cisco WebEx Connect clients; then you can click to call the person back and escalate to a web chat, video, or other multimedia session.

– You can access your messages using a Really Simple Syndication (RSS) feed.

• Cisco Unity Connection Web Inbox browser interface to voice messages

– You can view, sort, play, compose, forward, and reply to voice messages using the new HTML 5.0-based interface.

– You can easily address messages to multiple recipients and distribution lists.

– Voice messages in the Cisco Unity Connection Inbox are synchronized with the MWI on your telephone.

– Web browser-based tools are supported on Internet Explorer, Firefox, and Safari.

– The Web Inbox can be deployed as a widget or gadget.

• Internet Message Access Protocol (IMAP)-based email client to access voice messages

– You can access email and voicemail messages and play and delete voice messages from within the same desktop email client using the built-in ViewMail for Outlook or ViewMail for Notes player.

– Voice messages in your email inbox are synchronized with the MWI on your telephone.

– Various standards-based desktop email clients are supported, including Microsoft Outlook, Lotus Notes, and Entourage for Mac.

– You can compose, reply to, and forward messages by using IMAP clients.

• Visual voicemail

– You can view, sort, play, compose, forward, and reply to voice messages from the screen on your IP phone without having to dial in to the system.

– You can view, sort, and play messages from within Microsoft Office Communicator and Microsoft Lync using the Cisco UC Integration™ for Microsoft Office Communicator and Cisco UC Integration™ for Lync, respectively.

Personal Web Administration

Cisco Unity Connection allows you to customize your personal settings from a web browser. You can quickly and easily establish or change personal settings such as your voicemail options, security codes, personal distribution lists, and message-delivery options. You can also use the web administration interface to define and manage personal call-transfer rules to customize the delivery of incoming calls based on caller, time of day, or calendar status.

Simplified Installation, Configuration, and Maintenance


Running on the Cisco Linux-based appliance platform, Cisco Unity Connection uses a common set of management and serviceability tools designed to provide a consistent experience and to streamline the ongoing management and operation of a Cisco Unified Communications System. A centralized Cisco Unity Connection 8.6 solution combined with the Cisco Unified Messaging Gateway (Version 8.5 and later) and a Cisco Unity Express (Version 8.5 and later) messaging solution at a branch-office location allows you to enable Cisco Unified Survivable Remote Site Voicemail. This solution uses Cisco Unity Express at the branch office to manage voice messages if the WAN to the centralized Cisco Unity Connection solution is down. Then, Cisco Unified Survivable Remote Site Voicemail restores voice messages to the central server when WAN service returns.

Virtualization

You can install Cisco Unity Connection 8.6 as a virtual machine on Cisco Unified Computing System™ (UCS), IBM, and HP platforms. Cisco Unity Connection 8.6 extends the virtualization support to include Fibre Channel (FC), Fibre Channel over Ethernet (FCoE), Small Computer System Interface over IP (iSCSI), and Network File Storage (NFS) storage area networks (SANs). Table 1 lists more features and benefits of Cisco Unity Connection 8.6.

Features and Benefits

New Features for Cisco Unity Connection Version 8.6(2)

• Unified messaging with Microsoft Office 365: • Voice messages are synchronized with the Office 365 inbox. • MWI and message status are synchronized. • Secure, private messages, mobile client, and calendar integration are all supported. • You can enable unified messaging for specific users or all users. • Mixed Office 365 and on-premises Microsoft Exchange environments are supported.

• The solution supports Cisco SpeechView Fully Automated (Standard) and SpeechView Human Assisted (Pro) service levels. • Mix Pro and Standard users on the same Cisco Unity Connection system • Cisco SpeechView supports unified messaging - voicemail transcriptions are synchronized with audio copies of messages when deployed in a unified messaging environment. • Supports secure messaging when used with Cisco Collaboration clients such as Cisco Jabber.

Message Access from the TUI

• 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 reverse, pause, or fast forward messages during 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 hear before playing a message that it has been sent to multiple recipients.

• You can be allowed to listen to all the recipients of the message.

• With the message locator, you can search for messages by caller ID, name, or extension in saved messages.

• You can record messages and mark them as regular, urgent, private, or secure.

• You can record messages and request a return receipt.

• You can record a live conversation with a caller and have the recording sent to your mailbox.

• You can switch between spelling name and extension when addressing a message.

• With live reply, you can immediately reply to messages from other users.

• You can access email messages over the phone using the TTS feature (for Microsoft Exchange 2003, 2007, and 2010).


• When TTS is enabled, a conversation tells you if the message has attachments; when an attachment is in a playable or readable format, the attachment is played or read.

• You can view, listen, respond to, and play back messages using the Cisco Unified Communications Widget for Visual Voicemail on Cisco Unified IP Phones.

• You can access the Microsoft Exchange calendar through speech or the TUI.

• You can browse the calendar and accept, decline, or cancel an Outlook appointment.

• If you inadvertently disconnect while sending a new message, replying to, or forwarding a message, and if the message has at least one recipient or a recording, Cisco Unity Connection can save the message as a draft and allow you to return to finish the message on a subsequent call.

• You can review and recall messages sent over a period of time.

• When you hang up or your call is disconnected, bookmarks allow you to call back into Cisco Unity Connection and resume listening to messages without losing your place.

Speech-Enabled Messaging*

• Speech Connect for Cisco Unity Connection, a speech-enabled Automated Attendant for the enterprise, allows you to connect quickly with your colleagues using only your voice (available with Cisco Unity Connection 7.1.3 and later).

• You can speak your voicemail password.

• You can speak dates and times.

• You can use speech commands to play and process messages (play, record, reply, forward, delete, save, etc.).

• You can use speech commands to edit and manage your personal greetings.

• You can use speech commands to address messages to private distribution lists.

• You can use speech commands such as pause, resume, speed up, slow down, skip ahead, and skip back to provide rich and granular control of messages and prompts.

• Speech-enabled directory handlers allow outside callers to use voice commands to reach Cisco Unity Connection users.

• You can temporarily use touch tones to change setup options, and then return to speech-recognition mode.

• A speech command tutorial is available.

• You can customize speech-enabled directory handler greetings.

* Speech-enabled messaging is available for U.S. English only.

Call-Transfer Rules

• You can define rules to route incoming calls by caller.

• You can define rules to route incoming calls by time of day.

• You can define rules to route incoming calls by your calendar free or busy status (Microsoft Exchange only).

End-User Features

• Single Sign-On for the Cisco Unity Connection browser applications is supported.

• If a call is dropped while you are recording a message, Cisco Unity Connection saves a draft message and you can continue recording where you left off during your next session.


• You can customize message-notification options, manage personal greetings, or change passwords with Cisco Unity Connection Assistant (the Cisco web browser-based personal administrator).

• You can select the conversation type: Full or brief prompts.

• You can record and then address a message, or address and then record a message.

• You can record a message for future delivery.

• You can record up to five personal greetings (alternative, busy, internal, off hours, or standard).

• You can manage an alternative greeting, set the expiration date or time, notify users when an alternative greeting is set, or require callers to listen to the full alternative greeting.

• You can forward calls directly to an alternative greeting (or other personal greeting) without ringing the phone.

• You can specify an after-greeting action; after a user greeting, callers can leave a message, sign in, or hang up, or they can be sent to call handlers, directory handlers, interview handlers, or other users.

• You can use flex stack to specify the order in which messages are presented over the phone: by urgency and then by last in, first out (LIFO) or first in, first out (FIFO).

• You can create private distribution lists and address messages to them through the TUI or GUI.

• You can provide message notification for new messages through devices such as Simple Mail Transfer Protocol (SMTP), Short Message Service (SMS), text pagers, and phone destinations.

• With a cascade message-notification feature, you can send additional notification types if a message is not retrieved.

• You can send notifications for messages from a particular user or phone number.

• You can select whether message counts are announced; totals, saved, and new counts are available.

• You can specify whether Cisco Unity Connection announces a transferred call.

• You can perform a supervised transfer for individual alternate contact numbers.

• You can view and play back messages using Visual Voicemail on Cisco Unified IP Phones. You can use soft keys on Cisco Unified IP Phones to access all messages, new messages, or messages from a specific subscriber or outside caller.

• You can use a RSS reader to retrieve voice messages.

• You can perform a "live reply" to someone who left a message from an external telephone.

• You can have a your voice messages synchronized with your Exchange Inbox (Single Inbox).

• With ViewMail for Microsoft Outlook (VMO) and ViewMail for IBM Lotus Notes (VMN) plug-ins, you can compose, reply to, forward, play, rewind, or pause messages directly from within the Outlook or Notes email client.

• You can compose, reply to, and forward messages by using IMAP clients.

• Through calendar integration with Cisco Unified MeetingPlace® 8.0, you can join a meeting that is in progress, hear a list of participants for a meeting, send a message to the meeting organizer or participants, and set up an immediate meeting.

• You can dispatch a message to a group, with the message being assigned to the first member of the group to listen to the message. When the message is assigned, it is deleted from all other users' inboxes and becomes a normal message in the assignee's mailbox.

• You have flexibility with support for partitions, search spaces, and search scopes.

• You can receive and forward fax messages through integration with supported fax servers.

• You can customize subject lines for messages received in any visual client that displays the subject message, such as Outlook or an IMAP or RSS client.


• You can use a single phone number for both voice calls and fax transmissions.

• You can receive message-aging alerts before messages are deleted from the system.

• Over the phone, you can toggle between touch-tone and speech-recognition conversations.

• With the Voice Message Store and Forward feature, administrators, on a per-user basis, can forward voice messages to an external mailbox, making it easier for you to access voice messages on a mobile device.

System Administration Overview

• Cisco Unity Connection supports digital networking for up to 100,000 users within an enterprise and up to 20 servers or active-active cluster server pairs, including cross-server login, cross-server transfer, and cross-server live replay.

• Cisco Unity Connection is scalable to 250 ports and 20,000 users per server. Refer to the Cisco Unity Connection Supported Platforms List for details at: http://www.cisco.com/en/US/products/ps6509/products_data_sheets_list.html.

• High-availability support is achieved through an active-active redundancy configuration, which also supports up to 500 ports in the server pair.

• You can use advanced Cisco Unity Connection to Cisco Unity networking to allow both solutions to be networked together transparently.

• Cisco Unity Connection supports the synchronization of user information using Lightweight Directory Access Protocol (LDAP) with Microsoft Active Directory 2000, 2003, and 2008; Sun One; Sun iPlanet; Netscape Directory Server; OpenLDAP; and ADAM/LDS, enhancing your deployment and administrative options.

• Cisco Unity Connection allows for separation of an active-active pair across data centers (geospatial separation), providing greater deployment options for the enhanced reliability of high availability across the WAN.

• Cisco Unity Connection supports Voice Profile for Internet Messaging Version 2 (VPIMv2), which allows networking of up to 50 Cisco Unity, Cisco Unity Express, or third-party voicemail systems, allowing users on each of these systems to transparently reply to, forward, and exchange voice messages.

• Cisco Unity Connection 8.5 and later supports unified messaging (Single Inbox) with Microsoft Exchange 2003, 2007, and 2010, Microsoft BPOS-D, and Office 365.

• Support for virtual machine deployments on Cisco UCS, IBM, and HP servers and blades is specifications-based. Virtualization support is extended to include FC, FCoE, iSCSI, and NFS SANs.

• Phone-system integrations include any phone system that provides a serial data link (Simplified Message Desk Interface [SMDI], Message Center Interface [MCI], or Message Digest Algorithm 110 [MD110] protocol) to the master PBX IP media gateway (PIMG) unit (serial integration through analog PIMG or T1 IP media gateway [TIMG] units).

• Cisco Unity Connection integrates with QSIG-enabled PBXs through either Cisco Unified Communications Manager or a Cisco Integrated Services Router.

• Cisco Unity Connection integrates with Cisco Unified Communications Manager and leading traditional telephone systems, even simultaneously (using the PIMG or TIMG).

• Cisco Unified Communications Manager 4.1(3) and later, Cisco Unified Mobility Advantage, and Cisco Unified Mobile Communicator are supported.

• Cisco Unity Connection 8.5 and later supports the Cisco Unified Communications Manager Session Management Edition 8.5 and later.

• Integrations of Cisco Unity Connection 8.0 and later and Cisco Unified Communications Manager Express 3.4 are supported when the Cisco Unified Communications Manager Express 3.4 is installed on Cisco IOS® Software Release 12.4(6)T1 or later.

• Cisco Unity Connection 8.5 and later supports the Cisco Unified SIP Proxy integration.

• Cisco Unity Connection provides a browser-based system administration console and tools for easy installation and maintenance.

http://www.cisco.com/en/US/products/ps6509/products_data_sheets_list.html


• Cisco Unity Connection provides fax integrations with the Cisco Fax Server (Version 9.0 and later), OpenText Fax Server (RightFax Editiion Version 9.0 and later), and Sagemcom Xmedius Fax SP (Version 6.5.5).

• City and Department fields are available for administratively defined contacts.

System Administration Features

• The solution supports E.164 formatted phone numbers.

• Alternate extensions are configurable by the system administrator or user.

• Alternate key mappings for message retrieval can help you transition from traditional voicemail systems.

• Custom keypad mapping allows administrators to create TUIs for specific user needs. Call-routing rules can be configured to route calls to different conversation styles.

• Automatic gain control provides consistent message volume playback levels.

• Handlers provide building blocks for Automated-Attendant and intelligent call-routing functions. • Call handlers accept calls, play recorded prompts, route calls, and accept messages. • Directory handlers manage the way that callers search the directory. • Interview handlers collect and record input from callers.

• You can customize directory handlers with a voice greeting.

• You can configure per-user message-handling actions to determine how messages of specific types are handled in the system, such as "accept the message", "reject the message", or "relay the message".

• Caller ID is supported.

• Call screening is configurable.

• Class of service (CoS) controls user access to features.

• Administrators can create users individually or in bulk.

• Administrators can update users and districbutions lists and their various setting in bulk.

• Adminstrators can create and updated multiple alternate first and last names for contacts and users.

• Administrators can import users from Cisco Unified Communications Manager.

• Messages are day- and time-stamped.

• You can perform a directory search by spelling a username; you can enter up to 24 letters.

• You can log in to the TUI without entering your ID.

• Representational State Transfer-based application programming interfaces (APIs) for provisioning, messaging, telephony, and notifications allow integrations with existing corporate provisioning tools or messaging clients.

• Cisco Unity Connection 8.5 and later supports IPv6 addressing with Cisco Unified Communications Manager (7.1(2) or later) phone system integrations using Skinny Client Control Protocol (SCCP) and Session Initiation Protocol (SIP). The addressing mode is configurable by port group.

• Encrypted SCCP, Secure Real-Time Transport Protocol (SRTP), and Transport Layer Security/SRTP (TLS/SRTP) for SIP facilitates Cisco Unified Communications Manager integration.

• SIP support includes the following: • TLS/SRTP: Cisco Unified Communications Manager SIP trunk integrations support authentication and encryption of the Cisco Unity Connection voice messaging ports. • Keypad Stimulus Protocol (KPML): For Cisco Unified Communications Manager SIP trunk integrations, administrators can configure the integration to send dual-tone multifrequency (DTMF) keystrokes in the Real-Time Transport Protocol (RTP) media stream (in-band) or in a SIP message (out-of-band). • Port multiplexing: SIP integrations (such as for PIMG, TIMG, or Cisco SIP Proxy Server) can share the same SIP port on the Cisco Unity Connection server.


• Simple Network Management Protocol (SNMP) Versions 1, 2, and 3 are supported.

• Event logging is supported.

• Cisco Unity Connection 8.5 and later support application and database audit logging that tracks changes to the system in separate audit log files.

• Full mailbox warning is supported.

• You can create folders within a mailbox for inbox, deleted items, sent items, and draft items.

• Users can be set up for unified (single Inbox) or integrated (IMAP) messaging. These features can be enabled for individual users.

• Installation is simple and quick.

• A list of observed holidays is configurable.

• You can configure how Cisco Unity Connection handles messages that are interrupted by disconnected calls.

• MWI is supported, including enhanced MWI that displays a constant message count on certain Cisco Unified IP Phones.

• Multiple administrative levels allow you to control access to pages in the system administration GUI by CoS (read, modify, or delete rights).

• Music on hold (MOH) is supported.

• Nondelivery or delivery receipt reason details are presented in the GUI inbox.

• You can specify the public distribution lists to which new users will be added.

• Restriction tables are configurable.

• You can exclude return receipts.

• The system schedule is configurable.

• Self-enrollment allows you to set your password, record your voice name, and specify your directory listing.

• A status monitor allows for real-time administrator status of telephone ports, reports in progress, and system configuration.

• System broadcast messages for officewide announcements are supported.

• System greetings are configurable.

• The system offers 12- and 24-hour clock support for time stamps.

• The system time clock adjusts automatically for Daylight Savings Time.

• A TUI greetings administrator (Cisco Unity Connection Greetings Administrator) is supported.

• LDAP directory integration allows users to be quickly imported, synchronized, and authenticated within the directory.

• You can create up to nine mailbox stores in addition to the default mailbox store that is created when Cisco Unity Connection is installed.

• The system can be set up to shred voice messages for secure deletion.

• Message recording expiration guarantees voice messages cannot be listened to after they reach a set expiration date.

• Message aging alerts are supported.

• You can simulate abbreviated extensions by using prepended digits for call handlers and user mailboxes.


Security

• Cisco Unity Connection uses Security-Enhanced Linux (SELinux) access-control policies to provide a secure system.

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

• Call-restriction tables to prevent toll fraud are supported.

• Security event logging and reports of failed login and account lockouts to help prevent unauthorized PIN use are supported.

• Secure, private messaging prevents the playing of private messages accidentally forwarded outside the enterprise.

• A message-aging policy for secure messages automatically deletes all secure messages that are older than the specified number of days.

• The system can be set up to shred voice messages for secure deletion.

• Message recording expiration guarantees voice messages cannot be listened to after they reach a set expiration date.

• Message-aging policies can be set on a per-user basis.

• Secure RTP and signaling encryption provides for secure communication between Cisco Unity Connection and Cisco Unified Communications Manager.

• A user telephone PIN reset feature in Cisco Unity Connection Assistant reduces help-desk calls and operating expenses.

• Support for Secure HTTP (HTTPS) provides for secure web access to Cisco Unity Connection and allows for playback of secure messages within Microsoft Outlook.

Voice Quality

• G.722 and Internet Low Bitrate Codec (iLBC) voice codecs are supported (advertised or "on the line"). G.711 mu-law, G.711 a-law, and G.729 are also supported.

• System-level recording is available for linear pulse code modulation (PCM), Global System for Mobile Communications (GSM) 6.10, G.711 mu-law, G.711 a-law, G.729a, and G.726 through system-based transcoding resources.

Reports

• Call Handler Traffic Report

• Distribution Lists Report

• Events Report

• Outcall Billing Report

• Port Usage Report

• Users Report

• User Message Activity Report

• System Configuration Report

• Transfer Call Billing Report

• User Access Activity Report


• User Lockout Report

• Message Traffic Report

• Port Activity Report

• Mailbox Store Report

• Dial Plan Report

• Dial Search Scope Report

• For a full list and description of reports, refer to the Cisco Unity Connection System Administration Guide at: http://www.cisco.com/en/US/products/ps6509/prod_maintenance_guides_list.html.

Localization

The Cisco Unity Connection TUI, end-user GUI, and TTS engine are available in the following languages:

• Arabic (no TTS) • Catalan • Chinese (Hong Kong, Mandarin TUI with simplified and traditional Chinese GUI, simplified Mandarin TTS, but no traditional Mandarin TTS) • Czech • Danish • Dutch • English (U.S., U.K., and Australian) • English TTY • French (European and Canadian) • German • Greek • Hebrew (no TTS) • Hungarian • Italian • Japanese • Korean • Norwegian • Polish • Portuguese (Brazilian and European) • Russian • Spanish (European and Latin American) • Swedish • Turkish (no TTS)

2.3.7 Describe the maintenance and administration for all sites of the system.

Response: The Cisco Unified Communications Manager Real-Time Monitoring Tool (RTMT),

which runs as a client-side application, uses HTTPS and TCP to monitor system performance,

device status, device discovery, and CTI applications for Cisco Unified Communications

Manager. RTMT can connect directly to devices via HTTPS to troubleshoot system problems.

RTMT allows you to perform the following tasks:

Monitor a set of pre-defined management objects that monitor the health of the system

Generate various alerts, in the form of emails, for objects when values go over/below

user-configured thresholds

Collect and view traces in various default viewers that exist in RTMT

Translate Q931 messages

View syslog messages in SysLog Viewer

http://www.cisco.com/en/US/products/ps6509/prod_maintenance_guides_list.html


Work with performance-monitoring counters.

After a solution is deployed, ongoing management and maintenance can quickly become

burdensome – especially in distributed environments. Because the Cisco Unified

Communications Manager can easily be monitored and managed from a remote location,

organizations can reduce staffing requirements, downtime, and the costs associated with onsite

visits.

The design of the Cisco Unified Communications Manager appliance facilitates the ability to

perform remote and automated change management. Instead of using special hardware or

software, administrators can manage and monitor the appliance securely through the Web using

HTTPS or Secure Shell (SSH) Protocol (a CLI accessed through a secure shell). To streamline

administrative tasks and reduce the overall time required to manage the installation and upgrade

process, administrators can take advantage of an unattended installation option and invoke

upgrades and determine software versions through the browser or CLI. The browser- and CLI-

based installations also eliminate the administrative overhead associated with manual reboots, re-

logins, disk swaps, and other administrative tasks.

Many organizations schedule upgrades during non-peak hours to avoid interrupted business

operations – a major inconvenience for the IT administrators who need to stay late or come in

over the weekend to manage the process. A unique dual-partition mechanism within the Cisco

Unified Communications Manager appliance allows administrators to perform software upgrades

on a standby disk partition while preserving the current version of the Cisco Unified

Communications Manager software. This dual-partition mechanism — combined with the ability

to easily and quickly revert to a previous version of the software — significantly reduces the

time, burden, and risk typically associated with upgrades. Organizations can either upgrade the

image after hours while the system is operational or perform the bulk of upgrade work during

normal business hours — all the while reducing the time spent performing the upgrade.

Just as important, the integrated nature of the appliance reduces the number of upgrades needed

to stay current, resulting in simplified management and less downtime. Organizations running

Cisco Unified Communications Manager in a non-appliance mode could potentially install

separate software multiple times per year to address each of the four components. With an

appliance model implementation, the software elements are bundled into a single package. As a

result, administrators deal with only one software element for each minor, major, or maintenance

update.

2.3.8 What remote service capabilities are supported by the system and how are they used to provide technical support by the vendor.

Response: Cisco Unified Communications Manager provides a rich feature set for monitoring,

diagnostic, and repair capabilities.

Monitoring:

Performance-monitoring SNMP statistics from applications to SNMP manager or to

operating system performance monitor.


Real-Time trace monitoring.

Analysis Manager: Analysis Manager is a feature of the Cisco Unified Communications

Manager Real-Time Monitoring Tool (RTMT) that allows RTMT to provide solution-

level diagnostic features for other Cisco Unified Communications applications.

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.

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

Cluster-wide trace-setting tool

Trace collection tool

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.

2.3.9 Explain how the system will scale to up to 20% additional user capacity and how additional sites are added to the system.

The core unified communications system can scale to 30K devices per cluster. The current design would allow for a 25% or more scaling on the current hardware, if scaling is required beyond the 25% additional UCS hardware can be added to accommodate additional users. Additional sites are added in software, hardware may be required if existing site does not have switches or routers that support SRST.

2.3.10 Describe the installation process and provide references on the installation process.

Response: To Be Provided by the Partner


2.3.11 Explain the network requirements for supporting the proposed system to deliver high quality voice to both local and remote sites.

High quality voice and video for local and remote sites relies heavily on quality of service embedded in the network. Quality of service ensures that network resources are prioritized. The exact requirements vary based on design, but typical WAN requirements vary from 12k – 30k of bandwidth per voice call, LAN calls typically vary from 80k – 120k per voice call

2.3.12 Describe the required or recommended training for system administrators and end users for the system including time and costs.

Response: Admin Training times will vary by University based on IT Staff familiarity with Cisco Solutions etc. As for what is trained on please see training documents for information on both admin and end user training designs

3 IP COMMUNICATIONS SYSTEM SOFTWARE AND HARDWARE

3.1 System Software

3.1.1 Which software package is being proposed? Please provide the release and version?

Response: This proposal includes Cisco Unified Workspace Licensing is provides the latest versions of the following applications:

Cisco Communications Manager for call control (Version 8.6)

Cisco Unity Connection for voice messaging (Version 8.5)

Cisco Unified clients, mobility (Version 8.5)

Cisco Unified Presence (version 8.5)

3.1.2 Describe all the system software components for call process and identify the platforms where they are hosted in the proposed architecture.

Response: The Cisco Unified Communications Manager will provide the call control component in the proposed solution. 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. Cisco Unified Communications manager is deployed in a clustered fashion, allowing multiple servers to work together to provide a redundant solution. A publisher and one or more subscribers comprise a cluster that provides call processing functionality. The solution provides flexibility of deployment, allowing it to be installed on MCS servers, UCS servers, or third party servers (HP or IBM).


3.1.3 Identify how the proposed software maintains call processing services to the users at all sites during server or WAN failures.

Response: The proposed Cisco Unified Communication Manager allows for deployment that utilizes a cluster approach with up to eight call processing servers per Cisco Unified CM cluster. Cisco Unified Communication Manager groups devices (voice gateways, IP Phones, and CTI Ports) into device pools and allows for device pools to have a primary, secondary, and tertiary Cisco Unified Communication Manager server. When a device pool’s primary Cisco Unified Communication Manager server fails, the devices within that device pool automatically fail over to the secondary or tertiary Cisco Unified Communication Manager server. This type of redundancy may cross technological boundaries. For example, a phone may have as its first three preferred call control agents, three separate Unified Communication Manager servers belonging to the same call processing cluster. As a fourth choice, the phone can also be configured to rely on a Cisco IOS router for call processing services. Cisco Unified CM also provides geographical diversity and redundancy through support of a distributed deployment model which allow for nodes to be deployed nodes across WAN links. This means that if an entire site is off-line (such as would be the case during an extended power outage exceeding the capabilities of provisioned UPS and generator backup systems), another site in a different location can ensure business continuity.

3.2 Hardware Configuration

3.2.1 What hardware is being proposed? Please provide the model name and number.

Response: Cisco UCS C210 Rack-Mount Server - (http://www.cisco.com/en/US/products/ps10889/index.html)

3.2.2 Describe the IP call processing hardware platform in detail. Is it based on industry standard hardware, or is it proprietary?

Response: The Cisco UCS C210 server is a general-purpose , industry-standard form factor and x86 architecture, two-socket, 2 RU rack-mount server. It is designed to balance performance, density, and efficiency for storage-intensive workloads. (Data Sheet: http://www.cisco.com/en/US/prod/collateral/ps10265/ps10493/data_sheet_c78-587522.html)

3.2.3 What is the maximum user capacity of the proposed IP communications system? Provide a description of how scalability is achieved.

Response: The system can scale to over 100K endpoints, scale is achieved through the addition of virtual servers to provide additional call processing capabilities.

http://www.cisco.com/en/US/products/ps10889/index.html

http://www.cisco.com/en/US/prod/collateral/ps10265/ps10493/data_sheet_c78-587522.html


In a virtualized deployment, most Unified Communications applications such as Unified CM must be installed using a predefined template that specifies the configuration of the virtual machine's virtual hardware. These templates are distributed through Open Virtualization Archives (OVA), an open standards-based method for packaging and distributing virtual machine templates. These OVA templates define the number of virtual CPU, the amount of virtual memory, the number and size of hard drives, and so forth, and they determine the capacity of the application. More Information: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/callpros.html#wp1156290

3.2.4 What is the maximum number of simultaneous conversations supported by the proposed system? Is the system non-blocking for voice calls?

Response: The system is non-blocking and con accommodate as many conversations as endpoints or more in terms of external and multi-party conferences. More Information: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/callpros.html#wp1287334

3.3 Network Infrastructure Requirements

Ohio University Ohio University’s wired network consists of Juniper EX series of network switches with 10/100/1000 Mbps interfaces providing 802.3af Power over Ethernet on each port. The wireless network includes Cisco access points, wireless 802.11a/b/g network that provides approximately 8 Million square feet of Wi-Fi coverage in indoor and outdoor spaces on the main campus as well as on five regional campuses and two satellite campuses. Any proposed system should include a network readiness assessment. This assessment will detail any network readiness that would be required for a successful deployment. Including but not limited to bandwidth, QoS, PoE, etc. Bowling Green State University Bowling Green State University’s network infrastructure is a Cisco network with redundant Cisco 6509s in the core and distribution layers. All core and distribution layer switches are connected at 10G. The access layer is made up of 130 4506 chassis in the Academic and Administrative buildings, and 450 3560 switches in Residence Halls and computer labs. The majority of switch ports are 10/100, and there is some Power over Ethernet support, primarily for wireless access points and security cameras. Bowling Green State University has an extensive Cisco controller based wireless deployment. There are approximately 1,300 Cisco access points deployed providing coverage to over 85% of the University’s academic and administrative buildings and over 70% of the Residence Halls. Plans are in being developed to complete wireless in the remaining locations. Fourteen Cisco wireless controllers manage the wireless network. 70% of the wireless access points currently deployed are Cisco 1142s with 802.11 a/b/g/n radios. The other 30% are a variety of older Cisco 802.11 b/g access points.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/callpros.html#wp1156290

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/callpros.html#wp1287334


The network would need to be evaluated to ensure the access switching would meet the PoE and QOS requirement in all areas. Network upgrades shall be proposed by the vendor. Shawnee State University Shawnee State University has core distribution switching in place but lacks the POE access switching in some areas. The network would need to be evaluated to ensure the access switching would meet the POE and QOS requirement in all areas.

3.3.1 Describe requirements to the data network to support the system including necessary infrastructure features and capabilities.

Response: Campus LAN infrastructure design is extremely important for proper Unified Communications operation on a converged network. Proper LAN infrastructure design requires following basic configuration and design best practices for deploying a highly available network. Further, proper LAN infrastructure design requires deploying end-to-end QoS on the network. These required LAN infrastructure requirements are discussed in detail in the Cisco Solution Reference Network Design Guide, located here: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/netstruc.html#wp1043617

3.3.1.1 What capabilities are required inside the LAN?

Response: The deployment of an IP Communications system requires the coordinated design of a well structured, highly available, and resilient LAN infrastructure as well as an integrated set of network services including Domain Name System (DNS), Dynamic Host Configuration Protocol (DHCP), Trivial File Transfer Protocol (TFTP), and Network Time Protocol (NTP). Unified Communications places strict requirements on IP packet loss, packet delay, and delay variation (or jitter). Therefore, you need to enable Quality of Service (QoS) mechanisms into the LAN. Redundant devices and network links that provide quick convergence after network failures or topology changes are also important to ensure a highly available infrastructure. Additional information can be found at the following link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/netstruc.html#wp1043617

3.3.1.2 What capabilities are required across the WAN?

Response: Because voice is typically deemed a critical network application, it is imperative that bearer and signaling voice traffic always reaches its destination. For this reason, it is important to choose a WAN topology and link type that can provide guaranteed dedicated bandwidth. Cisco highly recommends that you design your Unified Communications network to conform to the following strict requirements: • Average IP packet loss <= 1%

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/netstruc.html#wp1043617



• Average delay variation (jitter) <= 30 ms • Average one-way packet delay <= 150 ms Additional information can be found at the following link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/netstruc.html#wp1184387

3.3.2 How does your proposed intelligent network infrastructure support end-to-end QoS? In a converged network supporting voice and data, how are QoS issues resolved?

Response: For applications such as voice, packet loss and delay results in severe voice quality degradation. Therefore, QoS tools are required to manage buffers and to minimize packet loss, delay, and delay variation (jitter). The following types of QoS tools are needed from end to end on the network to manage traffic and ensure voice quality: • Traffic classification Cisco IP Phones mark voice control signaling and voice RTP streams at the source, Cisco recommends the following classifications as best practices in a Cisco Unified Communications network.

• Voice: Voice is classified as CoS 5 (IP Precedence 5, PHB EF, or DSCP 46). • Videoconferencing: Videoconferencing is classified as CoS 4 (IP Precedence 4, PHB AF41, or DSCP 34). • Call signaling: Call signaling for voice and videoconferencing is now classified as CoS 3 (IP Precedence 3, PHB CS3, or DSCP 24) but was previously classified as PHB AF31 or DSCP 26.

• Queuing or scheduling By enabling QoS on campus switches, you can configure all voice traffic to use separate queues, thus virtually eliminating the possibility of dropped voice packets when an interface buffer fills instantaneously. Cisco recommends always using a switch that has at least two output queues on each port and the ability to send packets to these queues based on QoS Layer 2 and/or Layer 3 classification. • Bandwidth provisioning The addition of voice traffic onto a converged network does not represent a significant increase in overall network traffic load; the bandwidth provisioning is still driven by the demands of the data traffic requirements. The design goal is to avoid extensive data traffic congestion on any link that will be traversed by telephony signaling or media flows. Additional information can be found at the following link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/netstruc.html#wp1044009 http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/netstruc.html#wp1044317





3.3.3 Explain how you can provide easy addressing of the IP phones without having to change the addressing scheme of the existing IP data network?

Response: Proper access layer design starts with assigning a single IP subnet per virtual LAN (VLAN). When you deploy voice, Cisco recommends that you enable two VLANs at the access layer: a native VLAN for data traffic and a voice VLAN. Separate voice and data VLANs are recommended for the following reasons: • Address space conservation and voice device protection from external networks. Private addressing of phones on the voice or auxiliary VLAN ensures address conservation and ensures that phones are not accessible directly through public networks. PCs and servers are typically addressed with publicly routed subnet addresses; however, voice endpoints may be addressed using RFC 1918 private subnet addresses. • Ease of management and configuration. Separate VLANs for voice and data devices at the access layer provide ease of management and simplified QoS configuration. Additional information can be found at the following link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/netstruc.html#wp1043617

3.3.4 Explain how IP phones that are installed on the IP network are identified and added to the system?

Response: The following are the steps that a phone goes through once it has power: 1. The phone boots using its firmware. 2. The VLAN is determined. If a voice VLAN has been configured, the phone will learn that VLAN otherwise it will use the access VLAN. 3. If the phone is using DHCP, a DCHP request is sent. When the phone recieves the DHCP information, it will also receive the IP address of the TFTP server. 4. Once the phone learns the TFTP server address, it requests a configuration file. Since every phone has a unique configuration, the MAC address is used as part of the configuration file name. 5. If the phone has previously been configured within Communications Manager, a configuration file will exist in the TFTP server and will be sent to the phone. 6. Once the phone has loaded its configuration file, it will complete the registration with Communications Manager. Additional information can be found at the following link: http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7965g_7945g/6_1_3/english/administration/guide/7965str.html#wp1097038


http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7965g_7945g/6_1_3/english/administration/guide/7965str.html#wp1097038

http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7965g_7945g/6_1_3/english/administration/guide/7965str.html#wp1097038


3.3.5 Can IP phones share existing Ethernet ports with data devices, or do the IP phones require additional Ethernet ports be added by the customer to support voice?

Response: Both phone and PC may share the same Ethernet link but utilizing the built in switch port in the IP Phone. This minimizes the amount of network infrastructure required for deploying IP-based Unified Communications. Logical separation between voice and data networks is implemented using separate VLANs for these two services. (http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/netstruc.html#wp1043629)

3.4 PSTN, Legacy and SIP Integration Interfaces

3.4.1 Identify all types of PSTN interfaces or trunks supported by the system.

Response: Using Cisco ISR routers we can support the following PSTN TDM trunks/interfaces: T1/E1 PRI BRI T1/E1 CAS and R2 Analog (FXO) Analog DID Analog CAMA (http://www.cisco.com/en/US/partner/products/ps10537/products_relevant_interfaces_and_modules.html)

3.4.2 If PRI is supported, identify supported protocols and PRI services such as ANI, DNIS, Caller ID Name and Number.

Response: We support all the services offered by the standard Q.931 protocol – ANI DNIS Caller ID (name and number) Redirecting Number information (RDNIS)

3.4.3 Identify all supported interfaces for integration with existing or legacy telephone equipment such as PBX’s, key systems, fax servers, etc.

Response: We can integrate with existing or legacy telephone systems using the following interfaces” T1/E1 PRI T1/E1 QSIG T1/E1 CAS and R2 BRI


Analog (FXO) trunk ports E&M Analog station (FXS) port (http://www.cisco.com/en/US/partner/products/ps10537/products_relevant_interfaces_and_modules.html)

3.4.4 Please describe how the solution integrates with SIP trunking, include required architecture and any/all design elements. Also detail how the solution ensures toll quality voice paths via SIP and how troubleshooting/monitor can be accomplished.

Response: The SIP trunk features available in the current release of Unified CM make SIP the preferred choice for new and existing trunk connections. The QSIG over SIP feature provides parity with H.323 intercluster trunks and can also be used to provide QSIG over SIP trunk connections to Cisco IOS gateways (and on to QSIG-based TDM PBXs. SIP OPTIONS ping provides dynamic reachability detection for SIP trunk destinations, rather than per-call reachability determination. SIP Early Offer support for voice and video calls can reduce or eliminate the need to use MTPs and allows voice, video, and encrypted calls to be made over SIP Early Offer trunks. SIP trunk normalization and transparency improve native Unified CM interoperability with and between third-party unified communications systems and SIP PSTN ServiceProviders. Normalization allows inbound and outbound SIP messages and SDP information to be modified on a per-SIP-trunk basis. Transparency allows Unified CM to pass SIP headers, parameters, and content bodies from one SIP trunk call leg to another, even if Unified CM does not understand or support the parts of the message that are being passed through. Although SIP trunks are supported natively in Unified Communications Manager it is common practice to use a Cisco Unified Border Element (CUBE) sometimes called a Session Border Controller, to provide for demarcation between Service Provider and Enterprise voice networks. Additionally, CUBE can provide SIP trunk redundancy and scalability, DTMF interworking and normalization features that often are required when using differing vendors SIP implementations. (http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.html#wp1144139)

3.5 Proposed System Cabling

3.5.1 Describe the system cabling including the number of wire pairs of wires or network connections required to support the specific hardware configuration, telephones, PSTN interfaces, and connections to legacy equipment.

Response: Account Team


3.6 Station hardware

3.6.1 Does the system support the following types of user equipment?

Equipment YES NO OPTIONAL

Analog Telephones (2500 Type) X

IP Telephones X

Proprietary Digital Phones X

Modems X

Fax Machines X

3.6.2 Provide a description of each analog telephone provided with the proposed system

Response: Cisco does not provide or manufacture Analog phones. Any third party analog phone will work with the proposed solution providing it connects into an FXS port, either via a Cisco router, Cisco VG224, or Cisco ATA. http://www.cisco.com/en/US/products/hw/gatecont/ps2250/index.html http://www.cisco.com/en/US/products/ps11026/index.html

3.6.2.1 Please specify the power requirements for each analog telephone and if they require local or closet power. On power failure, is the telephone disabled or are support services such as LCD/LED devices disabled?

Response: Analog telephones are subject to the same power requirements of LAN Switches and the devices which power them. Redundant power supplies should be provided for any mission critical device which powers analog devices for the entire LAN. Cisco offers on certain router modules a power fail safe, which provides a direct power fail line to an analog endpoint. The FXO Power Failure feature is a hardware feature built into the FXO cards that allows connectivity to an analog phone patched into the right pair of wires to be activated by a relay if power to the Cisco router containing the NM-HDA fails. This allows PSTN calls to be made via the FXO line normally connected to the router from a designated "red" phone in the office while power is out. For an FXO EM in slot 0 of the NM-HDA, the analog phone must be connected to wire-pair 14 (counting from 1 on the RJ-21 connector) to take advantage of the FXO Power Failover feature. For an FXO EM in slot 1 of the NM-HDA, the analog phone must be connected to wire-pair 24 (counting from 1 on the RJ-21 connector) to take advantage of the FXO Power Failover feature. If both EMs are populated with FXO modules, two "red" phones can be used. http://www.cisco.com/en/US/prod/collateral/modules/ps2797/prod_qas0900aecd8017014a_ps5855_Products_Q_and_A_Item.html

3.6.2.2 Are headsets available for all analog telephones?

Response: Cisco does not provide or manufacture Analog phones, or the headsets for the analog telephones in this question. Any third party analog phone will work with the proposed solution providing it connects into an FXS port, either via a Cisco router, Cisco VG224, or Cisco ATA. http://www.cisco.com/en/US/products/hw/gatecont/ps2250/index.html http://www.cisco.com/en/US/products/ps11026/index.html

http://www.cisco.com/en/US/products/hw/gatecont/ps2250/index.html


http://www.cisco.com/en/US/prod/collateral/modules/ps2797/prod_qas0900aecd8017014a_ps5855_Products_Q_and_A_Item.html

http://www.cisco.com/en/US/prod/collateral/modules/ps2797/prod_qas0900aecd8017014a_ps5855_Products_Q_and_A_Item.html

http://www.cisco.com/en/US/products/hw/gatecont/ps2250/index.html



3.6.2.3 Does your analog station equipment provide the following features?

Cisco does not manufacture analog telephones; however, the Cisco Unified Communications Manager supports the

following capabilities on analog station equipment.

FEATURE YES NO OPTIONAL

Audio Volume Adjust X

Call Forward Busy X

Call Forward No Answer X

Call Forward All Calls X

Call Hold / Release X

Call Park / Pickup X

Call Transfer X

Call Waiting X

Calling Line ID Name and Number X

Make / Drop Conference X

Last Number Redial X

Multiple Calls Per Line Appearance X

Call Waiting Caller ID Name and Number X

Prime Line Select X

Privacy X

Ringer Pitch Adjust Function of handset

Ringer Volume Adjust Function of handset

Shared Extensions on Multiple Phones X

Single Button Retrieve X

Speakerphone Mute Function of handset

Speed Dial (Auto-Dial) X

3.6.2.4 What per-user configuration is required for each analog phone deployed or re-deployed in the system?

Response: The configuration will be 2 parts, 1st part is the configuration of the analog voice gateway, like VG200, 2nd part will be the configuration of the CUCM 8.5. I have attached the detailed configuration URL, please take it for reference. VG 200 analog gateway configuration: http://www.cisco.com/en/US/docs/routers/access/vg202_vg204/software/vg2_vg4_voip_ps2250_TSD_Products_Configuration_Guide_Chapter.html CUCM 8.5 analog gateway and phone configuration:

http://www.cisco.com/en/US/docs/routers/access/vg202_vg204/software/vg2_vg4_voip_ps2250_TSD_Products_Configuration_Guide_Chapter.html

http://www.cisco.com/en/US/docs/routers/access/vg202_vg204/software/vg2_vg4_voip_ps2250_TSD_Products_Configuration_Guide_Chapter.html


http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_5_1/ccmcfg/b06gtway.html#wp1452041

3.6.3 Provide a description of each IP telephone available with the proposed system

Response: Cisco has many IP phone models: gray Vs colorful, 100Mbps Vs 1000Mbps, wired Vs wireless, video-enabled Vs audio-only, expensive Vs cheap, so customer has a lot of options. Overally, Cisco has 39XX, 69XX, 79XX, 89XX and 99XX IP phones, all of these phones can work together with the customer CUCM 8.5 (with CUWL). Here is the detailed information on Cisco website: 39XX: http://www.cisco.com/en/US/products/ps7193/index.html 69XX: http://www.cisco.com/en/US/products/ps10326/index.html 79XX: http://www.cisco.com/en/US/products/ps379/index.html 89XX: http://www.cisco.com/en/US/products/ps10326/index.html 99XX: http://www.cisco.com/en/US/products/ps10326/index.html

3.6.2.53.6.3.1 Please specify the power requirements for each IP telephone and if they require local or closet power. On power failure, is the telephone disabled or are support services such as LCD/LED devices disabled?

Response: Most of the Cisco IP phones support the POE(power over Ethernet),power injector and local power adaptor. Cisco strongly recommend customer use POE, with POE switch, customer doesn't need local power any more. For the power budget: 69XX: IEEE POE Class 1 and 2 (3.2W-5.8W) 79XX: IEEE POE Class 2 and 3 (5W-12W) 89XX: IEEE POE Class 1,2 and 3 (3.84W-15.4W) 99XX: IEEE POE Class 4 (15.4W) For the specific model, please check the datasheet URL:

39XX: http://www.cisco.com/en/US/products/ps7193/index.html 69XX: http://www.cisco.com/en/US/products/ps10326/index.html 79XX: http://www.cisco.com/en/US/products/ps379/index.html 89XX: http://www.cisco.com/en/US/products/ps10326/index.html 99XX: http://www.cisco.com/en/US/products/ps10326/index.html If the power failure, the IP phones will be disabled.










3.6.2.63.6.3.2 Are headsets available for all IP telephones?

Response: Headset ports are available on most of the IP phones, but the headset will not come with the IP phones by default. Customer will buy the 3rd party headset later. Please visit the www.plantronics.com for detailed information. For the headset port: No: 3905,3911,6901,7911G,7936G,7937G; Yes: All other 69XX,79XX,89XX,99XX models;

3.6.2.73.6.3.3 Does your IP station equipment provide the following features?

The following features are supported on Cisco IP Phones registered to the Cisco Unified Communications Manager:



Call Forward Busy X

Call Forward No Answer X

Call Forward All Calls X



Call Transfer X

Call Waiting X


Make / Drop Conference X



Call Waiting Caller ID Name and Number X

Prime Line Select X

Privacy X

Ringer Pitch Adjust X

Ringer Volume Adjust X


Single Button Retrieve X

Speakerphone Mute X


3.6.2.83.6.3.4 What per-user configuration is required for each IP phone deployed or re-deployed in the system?

Response:

http://www.plantronics.com/


You can automatically add phones to the Cisco Unified Communications Manager database by using auto-registration or manually add phones by using the Phone Configuration windows.

By enabling auto-registration, you can automatically add a Cisco Unified IP Phone to the Cisco Unified Communications Manager database when you connect the phone to your IP telephony network. During auto-registration, Cisco Unified Communications Manager assigns the next available sequential directory number to the phone. In many cases, you might not want to use auto-registration; for example, if you want to assign a specific directory number to a phone

If you do not use auto-registration, you must manually add phones to the Cisco Unified Communications Manager database.

After you add a Cisco Unified IP Phone to Cisco Unified Communications Manager Administration, the RIS Data Collector service displays the device name, registration status, and the IP address of the Cisco Unified Communications Manager to which the phone is registered in the Phone Configuration window.

Before a Cisco Unified IP Phone can be used, you must use this procedure to add the phone to Cisco Unified Communications Manager. You can also use this procedure to configure third-party phones that are running SIP, H.323 clients, CTI ports, the Cisco ATA 186 Telephone Adaptor, or the Cisco IP Communicator. H.323 clients can comprise Microsoft NetMeeting clients. CTI ports designate virtual devices that Cisco Unified Communications Manager applications such as Cisco SoftPhone and Cisco Unified Communications Manager Auto-Attendant use.

For more information please check the below link. http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_5_1/ccmcfg/b06phone.html#wp1302807

3.6.33.6.4 Can telephones from third parties also be used with the proposed system? State the types of third party telephones supported and recommended sources.

Response:

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 Unified CM and Cisco Unified Communications Manager Express (Unified CME) SIP capabilities. Vendors include IPCelerate (unified client for education space), RIM (Blackberry 7270 wireless LAN handsets) and IP blue (Softphone). Cisco has also worked with third-party vendor Grandstream to test their Grandstream GXP 2000 to ensure interoperability.

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

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_5_1/ccmcfg/b06phone.html#wp1302807

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_5_1/ccmcfg/b06phone.html#wp1302807


enable third-party vendors to test and verify the interoperability of their endpoints with Unified CM and Unified CME.

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

3.7 System Reliability

3.7.1 How does the system provide reliability for voice services? Explain how it avoids any single point of failure.

Response: Unified Communications services offer many capabilities aimed at achieving high availability. They may be implemented in various ways, such as: •Failover redundancy For services that are considered essential, redundant elements should be deployed so that no single point offailure is present in the design. The redundancy between the two (or more) elements is automated. For example, the clustering technology used in Cisco Unified Communications Manager (Unified CM) allows for up to three servers to provide backup for each other. This type of redundancy may cross technological boundaries. For example, a phone may have as its first three preferred call control agents, three separate Unified CM servers belonging to the same call processing cluster. As a fourth choice, the phone can also be configured to rely on a Cisco IOS router for call processing services. •Redundant links In some instances, it is advantageous to deploy redundant IP links, such as IP WAN links, to guard against the failure of a single WAN link. •Geographical diversity Some products support the distribution of redundant service nodes across WAN links so that, if an entire site is off-line (such as would be the case during an extended power outage exceeding the capabilities of provisioned UPS and generator backup systems), another site in a different location can ensure business continuance. For detail information about redundancy and High availability for Cisco unified communications manager deployment please have a look on the below link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/models.html#wp1116121 http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/cmapps.html

3.7.2 Explain how the system reacts when the routers and switches fail. Can reliable dial tone and call routing be achieved without purchasing redundant network hardware?

Response: Non Comply, No reliable dial tone and call routing be achieved without purchasing redundant network hardware. During the switch failure Cisco IP phones which are directly connected to it cannot register with Cisco unified communications manager and the phones will be down.

http://www.cisco.com/

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/models.html#wp1116121

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/cmapps.html


During the router failure, which acts as voice gateway, Phones registering from outside (branch site) cannot register with CUCM and will be not operational. No outgoing calls via voice gateway router will be possible without redundancy. Internal calls within the site will be working fine.

3.8 System/Station/User features

For the following features, use the table to indicate their availability. Note if any of these features are optional or result in additional charges.


Answer/Answer Release X

Attendant or Operator Console X


Automatic Attendant X

Auto Echo Cancellation X

Automated Call-by-call Bandwidth Selection X

Automated Phone Installation Configuration X

Automatic Phone Moves X

Admission Control On WAN Usage X

Call forwarding (Off Premise) X

Call forwarding (Ring and/or No Answer) X

Call forwarding (Self Directed) X



Call Transfer X

Call Waiting X


Call waiting Caller ID Name and Number X

Conference Calling X

Dial by Name Directory X

Direct Inward Dialing X

Direct Outward Dialing (DOD) X

Distinctive Ringing (internal vs. external call) X

Distinctive Station Ringing Pitch X

Extension Dialing Between Locations X

IP-based Integrated Messaging X


Lowest Cost Trunk Selection X

Multi-Station Hunt Groups Spanning Locations X


Multiple Line Appearances X

PRI Protocol Support X

Ringer Pitch Adjust X

Ringer Volume Adjust X



Speakerphone Mute X


Station Monitoring or Busy Lamp Field Across all Locations

X

TAPI 2.1 X

Temporary Set Re-Assignment for Traveling Workers

X

Toll and Nuisance Number (900,976,970,550,540 exchanges) Restriction

X

Tone On Hold X

Visual Message Displays (All digital telephones) (name, extension, etc.)

X

TTY. TTD functionality X

3.9 Desktop Call Management – VoIP Softphones

3.9.1 Describes the system’s desktop call manager and the call control features supported from the user’s desktop computer.

Response: Cisco’s VoIP Softphone solution is called Cisco Unified Personal Communicator (CUPC). CUPC is an enterprise instant messaging client with click to call capability, group chat, drag and drop contacts to start conference calls, and media escalation. A full list of features can be found below:

Presence bubbles

Enterprise instant messaging

Presence and instant messaging federation

Customizable contact lists

Media escalation

Integrated voice and video telephony

IP Phone control

Conferencing

Communications History pane

Visual voice messages access For a complete list of features and definitions please see the link below. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6789/ps6836/ps6844/data_sheet_c78-647911.html

3.9.2 Does the desktop call manager application provide directory dialing across all locations in the system?

Response:

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6789/ps6836/ps6844/data_sheet_c78-647911.html


Yes. CUPC integrates with Microsoft Active Directory 2008 or 2003 and OpenLDAP 2.4 for corporate directory look ups from the desktop client.

3.9.3 Does the desktop call manager provide caller history or call log to archive the user’s telephone use?

Response: Yes, 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.

3.9.4 Does the desktop call manager provide call routing information for delivered calls identify how the caller reached the users though the proposed system?

Response: Yes. As referenced in the question above: 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. For more information on capabilities of Cisco Unified Communications Manager see the link below: http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps556/data_sheet_c78-652908.html

3.9.5 Does the desktop call manager provide searching and dialing of the users configure contacts from standard desktop personal information managers such as Microsoft Outlook?

Response: Yes, you can access the capabilities of Cisco Unified Personal Communicator from common desktop applications such as Microsoft Office 2007 and Microsoft Office 2010 (Outlook, Word, Excel, PowerPoint, and SharePoint), including presence light-up and click-to-communicate (call and instant messaging) capabilities. For Office 2010, you can use the popular contact-card click-to-communicate (call or IM) icons directly from the application. These services save time and streamline workflows by allowing you to see user availability and initiate communications such as personal and group voice, video, and chat sessions without having to switch between applications. For more information please see the link below: http://www.cisco.com/en/US/prod/collateral/voicesw/ps6789/ps6836/ps6844/data_sheet_c78-647911.html

3.9.6 Does the desktop call manager provide name match and display when received caller ID information matches information in the user’s personal information manger?

Response: No, Cisco Unified Communications manager does not match the number to the personal address book; instead caller ID is pulled from inbound call routing.






3.9.6.1 Are the matched names also displayed on the user’s telephone?

Response: Caller ID is pulled from inbound call routing, not the personal address book. If name matching does not occur within Active Directory through inbound call routing then only the number will be displayed to the phone.

3.9.7 Does the desktop call manager provide speed dialing of the user’s configured frequently called numbers?

Response: With Cisco Unified Personal Communicator users can Search both their personal contact list and their corporate directory from one easy-to-use interface to locate and initiate any form of communication with contacts quickly and simply.

Cisco Unified Communication Manager supports user-configured speed dial and call forward through web access for your physical endpoints/devices.

3.9.7.1 Are the configured speed dial entries also available on the user’s telephone?

Response: User configured contact lists set up by the user in Cisco Unified Personal Communicator are not directly available on the user’s telephone; however, user access to Cisco Unified Communications Manager , via a web portal, allows users to configure speed dial options that will show up on the user’s ip phone. For more information on the user options web portal: http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7931g/8_6/english/user_guide/31usrop.html#wp1009838 Alternatively, with Cisco Unified Personal Communicator users have the ability to control their desktop Cisco Unified IP Phone to make, receive, and control voice and video calls. Call-control capabilities such as call transfers, escalation to on-demand conferences, call parking and retrieval, call forwarding, and movement of active calls to other devices such as mobile phones are also available to meet your real-time communications needs.

3.10 E911 Services

Please describe how your proposed solution will integrate with existing e9-1-1 functionality of each university. A detailed design should accompany your RFP response in detailing E9-1-1 services. Ohio University Ohio University currently utilizes Pinnacle (Paetec) for E-911 – please describe how your solution will integrate with Pinnacle. Also please discuss how your solution’s IP offering will track the location of users and ensure the e911 database is update (wired and wireless). Cisco’s Emergency Responder (CER) does not integrate to Pinnacle for E-911 as emergency responder is its own E-911 solution. Tracking on CER is accomplished via several options, with 3rd party switches we track based on IP subnets. There are 3rd party solutions available that work with both Cisco and 3rd party infrastructure, if you would like to use Pinnacle for E-911 you would need to work with Paetec to determine interoperability.

http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7931g/8_6/english/user_guide/31usrop.html#wp1009838


Bowling Green State University Main campus, meets e911 compliance by setting the station display with a 4 digit building code followed by the room number of the location of the jack that the phone is plugged in to. 911 calls are routed to the on campus police dispatch office with the location in the display as well as a call logging workstation to automatically record the ID display and station number of the caller. Bowling Green State University, Firelands campus, routes calls out the local trunks to the local county sheriff’s office. Bowling Green State University, Levis Commons, routes emergency calls out the local trunks to local county sheriff’s office. Cisco’s Emergency Responder (CER) will integrate natively to the existing infrastructure on campus and provide tracking of IP phones to the port level, subnet level, and switch level. This allows for a more flexible solution than currently because if phones move or change location their port information is always in tack and does not require any changes. Routing of calls is based on administrator preferences, if you would like to route directly to PSAP’s or to on campus security it is configurable on a per site basis and can be easily changed at any time. Shawnee State University Shawnee State University currently utilizes PSALI software to input 911 data to Frontier. Also please discuss how your solution’s IP offering will track the location of users and ensure the e911 database is update (wired and wireless). Cisco’s Emergency Responder (CER) software will output the PSALI information to be uploaded to your local telco. Uploads can be accomplished via 3rd party software solutions or you can simply provide the output to the local provider. Wired tracking is based on subnet, port, and switch to provide tracking of IP phones. Wireless tracking other than subnet based is not possible within CER but if you would like geo location services 3rd party solutions are available.

3.11 System Administration

3.11.1 Describe the system administration tool(s) that are available to provide integrated administration of the system across all locations.

Response: In Cisco Unified Communications Manager Administration, use the Bulk Administration menu and the submenu options to configure entities in Cisco Unified Communications Manager through use of the Bulk Administration Tool. The Cisco Unified Communications Manager Bulk Administration Tool (BAT), a web-based application, performs bulk transactions to the Cisco Unified Communications Manager database. BAT lets you add, update, or delete a large number of similar phones, users, or ports at the same time. When you use Cisco Unified Communications Manager Administration, each database transaction requires an individual manual operation, while BAT automates the process and achieves faster add, update, and delete operations. http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/b10bat.html#wp1074364

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/b10bat.html#wp1074364

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/b10bat.html#wp1074364


3.11.2 Is the system administration application accessible from any workstation on the LAN /WAN?

Response: Yes, Cisco Communications Manager can be accessed via any computer on the network with a supported web browser, by an administrator with credentials into the system. This means a centralized system can be managed from virtually anywhere the network reaches.

3.11.3 Is the system administration application accessed through a standard web browser?

Response: The system administration can be accessed via a standard web browser .The following Cisco Unified Communications Manager applications support HTTPS: Cisco Unified Communications Manager Administration, Cisco Unity Connection Administration, Cisco Unified Serviceability, the Cisco Unified CM User Options, Trace Collection Tool, the Real Time Monitoring Tool (RTMT), and the XML (AXL) application programming interface. http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/b01intro.html#wp1052081

3.11.4 Can moves and changes be “batched”, that is can block copy changes can be made to a number of subscribers or class of service simultaneously?

Response: Yes. Batch Provisioning is available. Subscriber services may be ordered using the web interface on an individual basis for a single subscriber. However, when deploying a large number of services, it is often desirable to combine these together into a single batch, which can be scheduled to run at a later time. Cisco Unified Provisioning Manager permits a single batch to contain multiple types of orders: add, change, or cancel. It also permits multiple types of services to be specified in a single batch operation; for example, a batch can contain a combination of phone and voicemail additions or changes. Batches can also be created to manage dial plan settings or infrastructure configuration changes in Cisco Unified Communications Manager. Batches can be run immediately upon uploading to Cisco Unified Provisioning Manager or may be scheduled for execution at a later time. http://www.cisco.com/en/US/prod/collateral/netmgtsw/ps6491/ps6705/ps7125/data_sheet_c78-677750.html#wp9000122

3.11.5 Can administration of multiple remote sites be done through a centralized workstation? Is there any limit to how many workstations are supported?

Response: You access the Cisco Unified Communications Manager Administration program from a PC that is not the web server or has Cisco Unified Communications Manager installed. No browser software exists on the server. See the "Web Browsers" section for more information on browsing to the server. http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/b01intro.html#wp1022596

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/b01intro.html#wp1052081


http://www.cisco.com/en/US/prod/collateral/netmgtsw/ps6491/ps6705/ps7125/data_sheet_c78-677750.html#wp9000122

http://www.cisco.com/en/US/prod/collateral/netmgtsw/ps6491/ps6705/ps7125/data_sheet_c78-677750.html#wp9000122





3.11.6 How is security provided to prevent unauthorized access to the administration application? Can some administrative users be defined with “view-only” permissions?

Response: Access to the administration application is password protected. Only administrators with the proper credentials can access the admin application. Permissions may be restricted based on user roles, so that various levels of administrative access and permissions are possible.

3.11.7 Is there a limit to the number of administrators that can be logged on to the system at one time?

Response: No, there is no preset or hard limit on concurrent administrator logins.

3.11.8 Does the administrative application have on-line help? If yes, describe.

Response: Yes, the administrative application has both searchable and context-sensitive online help pages.

3.12 System Maintenance and Upgrades

3.12.1 Explain the back-up procedures for the system configuration and information and how the administrator would reload the data if needed to restore a previous configuration?

Response: The Disaster Recovery System (DRS), which can be invoked from Cisco Unified Communications Manager Administration, provides full data backup and restore capabilities for all servers in a Cisco Unified Communications Manager cluster. The Disaster Recovery System allows you to perform regularly scheduled automatic or user-invoked data backups.

The Disaster Recovery System performs a cluster-level backup, which means that it collects backups for all servers in a Cisco Unified Communications Manager cluster to a central location and archives the backup data to physical storage device.

DRS restores its own settings (backup device settings and schedule settings) as part of the platform backup/restore. DRS backs up and restores drfDevice.xml and drfSchedule.xml files. When the server is restored with these files, you do not need to reconfigure DRS backup device and schedule.

Please refer to the “Disaster Recovery System Administration Guide for Release 8.5(1)” for Backup and Restore scenarios and procedures at the link below,

http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/drs/8_5_1/drsag851.html#wp42856

3.12.2 How are customers provided future software releases? How is software upgrades performed?

Response: If the customer subscribe to Cisco Unified Communications Software Subscription (UCSS) service contract, customer will be able to request any future releases of Cisco UC system software and Cisco will mail the software media to the customer for upgrade to the latest releases. If without UCSS, customer will have to purchase the software upgrade product in order to receive the new software releases.


The Cisco Unified Communications Manager Upgrade Utility, a nonintrusive tool, detects the health of the servers in the Cisco Unified Communications Manager cluster before you perform an upgrade to Cisco Unified Communications Manager.

Before you perform an upgrade to a Cisco Unified Communications Manager version, download and run the latest version of the Cisco Unified Communications Manager Upgrade Utility, a nonintrusive tool that detects the health of the servers in the Cisco Unified Communications Manager cluster without changing the state of the system.

To verify that the server meets the minimum requirement for the Cisco Unified Communications Manager version to which you are upgrading, refer to the Cisco Unified Communications Manager Server Support Matrix. To obtain the most recent version of this document, go to

http://cisco.com/en/US/docs/voice_ip_comm/cucm/compat/cmhwcomp.xls

Please refer to “Cisco Unified Communications Manager Upgrade Utility 4.3(10)” link below for more details,

http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/upgrade/assistant/up4310tl.html

3.12.3 When system or station software updates are performed, must the system be shut down, or can these types of activities take place in an on-line environment?

Response: While not recommended by Cisco, the software upgrade (major) or updates (minor) of Cisco Unified Communications Manager system can be performed in an on-line environment without system being shutdown. The Cisco Unified Communications Manager Upgrade Utility, a nonintrusive tool, detects the health of the servers in the Cisco Unified Communications Manager cluster before you perform an upgrade to Cisco Unified Communications Manager.

Please refer to “Cisco Unified Communications Manager Upgrade Utility 4.3(10)” link below for more details,


3.12.4 During a system upgrade, explain how each component of the system is upgraded including estimate total time for upgrade for the proposed system and the estimated time each service or component is off-line.

Response: You can run the Upgrade Utility on only one server at a time. Running this utility takes approximately 1 to 60 (or more) minutes for the publisher database server. The time that it takes on the publisher database server depends on the size of the backup file. The utility takes approximately 1 to 5 minutes for each subscriber server. Please refer to the “Running the Utility” chapter in the Cisco Unified Communications Manager Upgrade Utility 4.3(10)” for detailed procedures and tasks of using the utility below, http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/upgrade/assistant/up4310tl.html#wp42356

http://cisco.com/en/US/docs/voice_ip_comm/cucm/compat/cmhwcomp.xls


http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/upgrade/assistant/up4310tl.html#wp42356

http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/upgrade/assistant/up4310tl.html#wp42356


3.13 System Monitoring and Diagnostics

3.13.1 Describe the diagnostic tools available for monitoring and maintaining the system’s performance.

Response: The Cisco Unified Real-Time Monitoring Tool (RTMT), which comes with any Cisco Unified Communications Manager system, runs as a client-side application, uses HTTPS and TCP to monitor system performance, device status, device discovery, CTI applications, and voice messaging ports. RTMT can connect directly to devices via HTTPS to troubleshoot system problems.

RTMT allows you to perform the following tasks:

• Monitor a set of predefined management objects that monitor the health of the system.

• Generate various alerts, in the form of e-mails, for objects when values go over/below user-configured thresholds.

• Collect and view traces in various default viewers that exist in RTMT.

• Translate Q931 messages.

• View syslog messages in SysLog Viewer.

• Work with performance-monitoring counters.

Please refer to the “Administration Guide for Cisco Unified Real-Time Monitoring Tool version 8.5(1)” below for more details, http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/service/8_5_1/rtmt/rtintro.html

3.13.2 What remote diagnostics are available? Can administrators see and access any alarms or alerts on the system from remote terminals?

Response: To supplement the management and administration of the Cisco Unified Communications Manager system, you can use remote serviceability tools. Using these tools, you can gather system and debug information for diagnostic help or remote troubleshooting. The tools can process and report on a collection of local or remote Cisco Unified Communications Manager configuration information. With customer permission, technical support engineers log on to a Cisco Unified Communications Manager server and get a desktop or shell that allows them to perform any function that could be done from a local logon session.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/service/8_6_1/admin/saintdes.html#wp1064170

3.13.3 Can the system be configured to notify the administrator of diagnostic events when they are remote or away from the system?

Response: CUCM provides Cisco Unified Real-Time Monitoring Tool that allows monitoring real-time behavior of components through RTMT. The system generates alert messages to notify administrator when a predefined condition is met, such as when an activated service goes from up to down. The system can send alerts as e-mail/epage..

http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/service/8_5_1/rtmt/rtintro.html


3.13.4 For each of the following system monitoring items listed below, respond with a “Yes” if the proposed IP PBX monitoring features can support the feature. If the answer is “No”, then state when you expect the IP PBX to be able to support this feature.

Feature YES NO Availability Date

Status of all trunking Yes Current Feature

Status of all call routing components Yes Current Feature

Integrated status of all locations Yes Current Feature

Status of individual stations (IP / Analog)

Yes Current Feature

Call usage reporting Yes Current Feature

WAN usage reporting Yes Current Feature

IP quality statistics reporting Yes Current Feature

Diagnostic events listing or reporting Yes Current Feature

Real-time traffic status Yes Current Feature

Status of all gateway ports Yes Current Feature

4 IMPLEMENTATION

4.1 Project Management

4.1.1 Project Plan - Bidders are required to supply a complete description of the key activities required for the installation of the proposed syste dates below would be changed based on each individual project and appropriate time required for each taskj

Response:

Task Name Start Finish Resource Names

Client Name Fri 4/30/10 Fri 4/30/10

Project Launch Meeting Fri 4/30/10 Fri 4/30/10 Project Team

Planning Fri 4/30/10 Fri 4/30/10

Equipment Ordered Fri 4/30/10 Fri 4/30/10 NETech Corporation

Resources Assigned Fri 4/30/10 Fri 4/30/10 NETech Corporation

Discovery of Project Scope Fri 4/30/10 Fri 4/30/10 Project Team

Project Contact List Identified and Communicated

Fri 4/30/10 Fri 4/30/10 NETech Corporation

Identify Milestones Fri 4/30/10 Fri 4/30/10 Project Team

Define Project Plan/Timeline Fri 4/30/10 Fri 4/30/10 NETech Corporation

Discovery Fri 4/30/10 Fri 4/30/10

Call Flow Discovery and Documentation Fri 4/30/10 Fri 4/30/10 NETech Corporation

User Data Defined and Provided to NETech Fri 4/30/10 Fri 4/30/10 Client X

Implementation Document Review and Approval

Fri 4/30/10 Fri 4/30/10 Project Team

Hardware Installation Fri 4/30/10 Fri 4/30/10


Install and Build CM Publisher Fri 4/30/10 Fri 4/30/10 NETech Corporation

Install and Build CM Subscriber 1 Fri 4/30/10 Fri 4/30/10 NETech Corporation

Install and Build CM Subscriber 2 Fri 4/30/10 Fri 4/30/10 NETech Corporation

Install and Build Unity Connection Server Fri 4/30/10 Fri 4/30/10 NETech Corporation

Install and Build Unity Connection HA Server Fri 4/30/10 Fri 4/30/10 NETech Corporation

Install and Configure Voice Gateway Fri 4/30/10 Fri 4/30/10

Location 1 Fri 4/30/10 Fri 4/30/10 NETech Corporation


Install and Configure VG224s Fri 4/30/10 Fri 4/30/10 NETech Corporation

Customer Provided Servers Ready for NETech

Fri 4/30/10 Fri 4/30/10

CTI Fri 4/30/10 Fri 4/30/10 Client X

Synapps Fri 4/30/10 Fri 4/30/10 Client X

Configuration and Testing Fri 4/30/10 Fri 4/30/10

Configure Communications Manager Fri 4/30/10 Fri 4/30/10 NETech Corporation

Configure Unity Connection Fri 4/30/10 Fri 4/30/10 NETech Corporation

Configure Voice Gateways Fri 4/30/10 Fri 4/30/10

Test PSTN Connectivity Fri 4/30/10 Fri 4/30/10 NETech Corporation

Integration to Nortel/Test Extension Dialing

Fri 4/30/10 Fri 4/30/10 Client X

Deployment Fri 4/30/10 Fri 4/30/10

Set, Test and Deploy IP Phones Fri 4/30/10 Fri 4/30/10 NETech Corporation

End User Training Fri 4/30/10 Fri 4/30/10 NETech Corporation

Cutover Fri 4/30/10 Fri 4/30/10

Location 1 Cutover Fri 4/30/10 Fri 4/30/10

Analog Cutover Fri 4/30/10 Fri 4/30/10 Client X

PSTN Cutover Fri 4/30/10 Fri 4/30/10 NETech Corporation

Location 2 Cutover Fri 4/30/10 Fri 4/30/10

PSTN Cutover Fri 4/30/10 Fri 4/30/10 NETech Corporation

Post Cutover Support Fri 4/30/10 Fri 4/30/10



Administration Training Fri 4/30/10 Fri 4/30/10 NETech Corporation

Final Documentation Presented to Client Fri 4/30/10 Fri 4/30/10 NETech Corporation

Project Closure Fri 4/30/10 Fri 4/30/10 NETech Corporation

4.1.2 Transparency - It is essential that the installation of the new system be as transparent as possible to the users. There should be no telephone service interruptions, no interim changes in dialing procedures, and no perceived degradation in the quality of service.

Response:


Netech will work with the individual Universities to create a design and migration plan which will serve to minimize the outages. This plan will utilize multiple circuits to accommodate a hardware rollover plan that will reduce the outage to zero.

4.2 Installation Requirements

4.2.1 Responsibility - The selected vendor is solely responsible for the complete turn-key engineering of the new voice communications system and all interconnecting facilities. Please describe the design and implementation services recommended for implementing the proposed solution.

Response:

Planning – Design – Implementation – Operate – Optimization

NETech approaches projects with the PDIOO – Plan, Design, Implement, Operate and Optimize

methodology. PDIOO is a repeatable and measurable process that enables NETech to provide the most

relevant and effective technology solutions.

Plan

Ordering

Staffing

Internal Launch Meeting (ILM)

Project Launch Meeting (PLM)

Design

Design Discovery Sessions

Design Deployment Plan (DDP)

Client Approval

Implement

Stage and install

Configure

Test

Train

Cutover

Day 1 Support

Operate

Final Documentation

Project Closure Meeting

NETech Maintenance

Document

What’s Next?

Optimize

NETech Technical Advisor

Evergreen Process


PDIOO is a repeatable and measurable process that enables NETech to provide the most relevant and viable technology solutions. It incorporates PMP best practices centered on complex technology solutions.

PLAN During the planning phase NETech will order all equipment identified during the Solution Sales process. NETech evaluates the scope of the project and matches engineer skills and availability with the requirements identified in the Statement of Work. Communication is essential throughout the life of the project and starts with a formal handoff between our presales team and delivery, thus ensuring quality knowledge transfer. We then replicate the knowledge transfer with the customer in a Project Launch Meeting to start the communication flow.

DESIGN Developing a detailed design is essential to reducing risk, delays, and the total cost of network deployments. A design aligned with business goals and technical requirements can improve network performance while supporting high availability, reliability, security, and scalability. Day-to-day operations and network management processes need to be anticipated. To add detail, NETech conducts planning workshops, and administers more detailed surveys. At this time, the team decides how to best meet requirements for the applications, support, backup, and recovery while focusing on the business needs and objectives. The final deliverable of this phase, a Design & Deployment Plan (DDP), documents the detail that is required to implement a solution that meets the customers’ requirements and objectives. The DDP is presented to the client for approval prior to implementation to ensure that the design is a direct reflection of the desired goal.

IMPLEMENT Introduces the solutions into the network and business processes. This phase is a full implementation of the technology solution based on the DDP from the Design phase. NETech will initiate a formal execution of a test plan during this phase prior to the cutover to the new systems. Training and knowledge transfer are very important to ensure the new solution is adopted by the user community and support team. NETech is focused on customer satisfaction and will schedule to be onsite for Day One support making the customer experience that much better.

OPERATE Starts the transition to an operational function. The NETech team achieves system stabilization and finalizes the status of all deliverables, documentation, training, and remaining deployments. This phase also includes the hand off to NETech Support for ongoing troubleshooting and full system support of the technology implemented. Network operations represent a significant portion of IT budgets, so it’s important to be able to reduce operating expenses while continually enhancing performance. Throughout the operate phase, a company proactively monitors the health and vital signs of the network to improve service quality; reduce disruptions; mitigate outages; and maintain high availability, reliability, and security. By providing an efficient framework and operational tools to respond to problems, a company can avoid costly downtime and business interruption. Expert operations also allow an organization to accommodate upgrades, moves, additions, and changes while effectively reducing operating costs.

OPTIMIZE

The process is complete; however, NETech continues to identify technology enhancements and process


improvements. A good business never stops looking for a competitive advantage. That is why

continuous improvement is a mainstay of the network lifecycle. In the optimize phase, a company is

continually looking for ways to achieve operational excellence through improved performance,

expanded services, and periodic reassessments of network value. Have business goals or technical

requirements changed? Is a new capability or enhanced performance recommended? As an organization

looks to optimize its network and prepares to adapt to changing needs, the lifecycle begins again

continually evolving the network and improving results.

4.2.2 Initial Work - Vendor will perform station reviews, data base preparation, and original program initializations.

Response: Netech will perform the appropriate station reviews with the relevant individuals and department heads and prepare the import using the information gleaned from the interviews. The final step will be to program the system to accommodate custom requirements that were uncovered by the station reviews.

4.3 Facility Requirements

4.3.1 Bidders must furnish all space, power, and environmental requirements for the proposed telephone system and optional voice messaging equipment.

Space – Provide the physical dimensions of the proposed equipment.

Cisco C210 Servers consume (2) two rack units of space Cisco 3925 Routers consume (3) three rack units of space Cisco VG224 Gateways consume (1) one rack unit of space

Power - All power requirements, including any special conditioning or grounding requirements.

Cisco C210 Servers consume a maximum of 570 watts of power Cisco 3925 Routers consume a maximum of 420 watts of power Cisco VG224 Gateways consume a maximum of 60 watts of power Cisco 7942, 7962 and 6921 phones are Class 2 802.3af which consume 7 Watts

Heat - Vendor must provide heat dissipation for proposed switch room and the recommended

safe temperature operating range for the proposed system.

One ton of cooling is equivalent to 3500 watts of heat dissipation. This is not a hard and fast rule, but as a rule of thumb we usually need to see 1 ton of cooling for every 2500 watts dissipated. This is normally augmented by the CRAC vendors’ ability to size the units for air flow and efficiency.

Floor Loading - Vendor must provide complete floor loading requirements.


Cisco C210 Servers produce 50.7 Lbs (23 Kg) of load Cisco 3925 Routers produce 60 Lbs (27 Kg) of load Cisco VG224 Gateways produce 11 Lbs (4.1 Kg) of load

4.4 Training

4.4.1 Requirements - The successful bidder is required to conduct end-user training on all premises listed in this RFP, tailored specifically to each university’s particular requirements (e.g., console operators, call-contact center agents, administrative assistants, and professional).

Response: Comply

4.4.2 Training Plan - Vendor will also provide a training program and training materials for designated personnel who will train future employees.

Response: Comply

4.4.3 Description - For each product application proposed, provide a detailed description of the training the vendor will provide.

Response: see attached training syllabus

5 VENDOR SERVICE

5.1 Maintenance and Warranty

5.1.1 A complete maintenance and warranty agreement must be included as part of the bidder's proposal to support the business 24 hours a day, 7 days a week. Please provide 1, 3 and 5 year models with your response.

Response: Comply

5.1.2 Defective Parts - During the warranty period and any subsequent maintenance agreement, any defective components shall be repaired or replaced at no cost .

Response: Comply


5.1.3 Maintenance Personnel - All system maintenance during the warranty period and under any maintenance agreements shall be performed by the successful bidding organization and at no additional cost other than those charges stipulated to maintain the warranty.

Response: Comply

5.2 Logistical Support

5.2.1 Bidder should identify the address of the vendor's local service centers and the number of service personnel trained on the proposed system.

Response: Netech Corporation 22 Engineers 4960 Blazer Parkway Dublin, OH 43017 Netech Corporation 29 Engineers 4595 Broadmoor SE, Ste. 190 Grand Rapids, MI 49512 Netech Corporation 39 Engineers 26800 Meadowbrook Road, Ste. 119 Novi, MI 48377 Netech Corporation 42 Engineers 12272 Hancock St. Carmel, IN 46032

5.3 Repair Response

5.3.1 Repair Commitment - The bidder must include a description of the bidder's repair commitment from time of trouble discovery through the time the trouble is cleared.

Response: see attached Cisco SMARTnet document

5.3.2 Response Time - A guaranteed response time of no more than 4 hours for all major system problems and a maximum of 24 hours response to other system problems.

Response: comply


Comply

5.3.3 Major/Minor Problems - Bidders must describe their definitions of major and minor problems.

Response: On all core server system components there is 5 years of 4 hour response included in the quote so all major system problems are covered by that service level. For all minor issues (local redundancy on gateways etc) not putting system in a down status would be considered minor and are covered by Cisco’s world class SMARtnet support on a 8x5xnbd service level

5.3.4 Replacement Time - Explain the amount of time required for full replacement of the central operating hardware/software of the system, assuming a suitable site exists for locating the replacement components.

Response: Assuming an appropriate backup of the virtualized guest operating systems associated with all the systems involved, a new system could be built and place in an operating status in as little as (1) one day. This is dependent on the existence of an appropriate facility and the hardware delivered to the site in a ready state for operation.

5.3.5 Emergency Installation - How long does it take trained personnel to install and load operating system software and database software, if a major disaster destroys the call processing component of the system?

Response: Assuming an appropriate backup of the virtualized guest operating systems associated with all the systems involved, a new system could be built and place in an operating status in as little as (1) one day. This is dependent on the existence of an appropriate facility and the hardware delivered to the site in a ready state for operation. 6 CONFIGURATION/PRICING Respondent must itemize all charges for individually identifiable components of the proposed IP communications system, including all associated installation, programming, and cabling. Bidder must include charges for all components required to connect all applications, such as, all design charges, telco interface charges, and training charges. See attached BOM’s with itemized pricing 7 VENDOR REFERENCES The vendor must submit five references. This information should include company name, contact, telephone number and size of system. Ideal references would be institutes of higher learning of similar size to Ohio University, Bowling Green State University and Shawnee State University. Response :See Attached References sheet


8 VENDOR PRESENTATIONS/PILOT/BAKE OFF Each vendor may be required to present their proposed solution to each campus, separately and present to the core design/research team separately. Each university, at their desire, may choose to perform a 90 pilot of the proposed solution. Each university, at their desire, may require a bake off of proposed solution(s). The bake off requirement(s) is to be determined however any bake off review should mirror any proposed solution that you expect to install at the university. Response: Comply

9 GENERAL

9.1 Shortlist

The Shared Services Project reserves the right to shortlist the Bidders on all of the stated criteria. However, The Shared Services Project may determine that shortlisting is not necessary.

9.2 Interviews

The Shared Services Project reserves the right to conduct interviews with all or some of the Bidders at any point during the evaluation process. However, The Shared Services Project may determine that interviews are not necessary. In the event interviews are conducted, information provided during the interview process shall be taken into consideration when evaluating the stated criteria.

9.2 The Shared Services Project reserves the right to make such additional investigations as it deems necessary to establish the competence and financial stability of any firm submitting a proposal.

9.4 Prior Experience

Experiences with The Shared Services Project and entities which evaluation committee members represent shall be taken into consideration when evaluating qualifications and experience.


9.5 Negotiations

The Shared Services Project reserves the right to conduct discussions with Bidders, and to accept revisions of proposals, and to negotiate price changes. During this discussion period, The Shared Services Project will not disclose any information derived from proposals submitted. Once an award is made, proposal documents are public record and will be disclosed upon request (see Additional Terms & Conditions, in the Instructions for Bidders Section, below).

9.7 Deadline & Delivery

OHIO University on behalf of this Shared Services Project will receive proposals through the OHIO University Sourcing Solution (found at https://sciquest.ionwave.net/prod/vendorregistration.aspx?vcid=527) until 2:00pm EST, September 27 2011. Firms submitting proposals will be responsible for delivery of the documents via the OHIO University Sourcing solution as outlined above. Any proposal received after the time and date specified, prepared or submitted, may not be eligible for consideration. Bidders are cautioned to write all descriptions and monetary amounts clearly so there is no doubt as to the intent and scope of the proposal. Erasures and other changes in the proposal must bear the signature or initials of the bidder. ALL PROPOSALS MUST BE SIGNED. Failure to provide this information may result in rejection of the proposal. The Shared Services Project reserves the right to reject any proposal not prepared and submitted according to the provisions herein outlined, and may reject any or all proposals. Any proposal may be withdrawn prior to the proposal due date by accessing the OHIO University Sourcing site at https://sciquest.ionwave.net/prod/vendorregistration.aspx?vcid=527.

IMPORTANT NOTE: The bidder is cautioned against last minute attempts to meet the due date & time and OHIO University will not be responsible for network outages, failure to register as a bidder, and other related internet malfunctions on the part of the bidder in submitting their bid. The bidder will receive an electronic confirmation of successful submission of the bid. Plan ahead. 10 QUESTIONS REGARDING THIS PROPOSAL

10.1 Formal Questions.

All questions are considered formal and may only be submitted via email to [email protected] . All responses will be returned via the OHIO University Sourcing Solution as well and the questions and the response added as a “Note” to this Request for Proposal at https://sciquest.ionwave.net/prod/vendorregistration.aspx?vcid=527 Registered bidders will receive email notifications of each “Note” or “Addendum” added to this RFP. Please note that the email will be delivered from “[email protected]. All questions should be directed to the [email protected] via email no later than September 6, 2011, at 2:00pm EST.

10.2 Note:

Email question to the OHIO University official listed above only. Do not include any other recipient on the question.

Email the question from a company email system – i.e., use only an email account provided by your firm. Do not email questions from a personal email account.

https://sciquest.ionwave.net/prod/vendorregistration.aspx?vcid=527


mailto:[email protected]



Clearly identify yourself, your company, and the RFP number, in the body of the email. Do not include any proposal information in the body of the e-mail. Proposal information should be only submitted via the OHIO University Sourcing portal at: https://sciquest.ionwave.net/prod/vendorregistration.aspx?vcid=527 Include the following text in your signed submission:

I understand that the OHIO University Procurement Services will take care to protect the confidentiality of my proposal prior to the bid opening date and time. I also understand and acknowledge that as a result of my choice, to attach the proposal to this email, confidentiality cannot be assured in the traditional manner (e.g., unbroken seals). I assume full responsibility for my choice to submit this proposal and for the lack of assured confidentiality inherent in that choice.

11 ADDITIONAL TERMS & CONDITIONS OF SUBMITTING A PROPOSAL

11.1 Costs incurred by bidder

The bidder, by submitting a proposal, agrees that any cost incurred by the bidder in responding to this RFP, or in support of activities associated with this request, are to be borne by the bidder and may not be billed to the Shared Services Project. The Shared Services Project will incur no obligation or liability whatsoever to anyone by reason of issuance of this RFP, or action by anyone relative thereto.

11.2 Signature & submission

Proposals must be dated, signed by an official authorized to bind the bidder to the terms of the proposal and submitted to the OHIO University Procurement Services in accordance with the terms and conditions of this RFP.

11.3 Obtaining clarification

All issues and questions raised in this RFP must be answered in full. Each bidder understands and agrees that it has a duty to inquire about and clarify any RFP issue that the bidder does not fully understand or believes may be interpreted in more than one way. Every attempt will be made to promptly answer all inquiries from each bidder.

11.4 Freight Terms

If applicable, all prices quoted are to be F.O.B. Destination. Unless clearly stated otherwise by the bidder, prices quoted will include all charges for transportation, packaging, etc., necessary to complete delivery on an F.O.B. Destination basis.

11.5 No Bid Requirement

If you are unable to submit a proposal, please date and sign the Terms & Conditions sheet, and indicate “NO BID”. Give a brief explanation, and return the sheet before the due date.



11.6 Contractual obligations

The contents of proposals submitted by the successful bidder will be considered contractual obligations to each institution upon award.

11.7 Sales Tax

The institutions participating in this Shared Services Project, as an instrumentality of the State of Ohio, are exempt from Ohio sales tax and Federal excise tax, including Federal transportation tax.

11.8 Formal & Informal RFPs/Proposal opening

This is an informal RFP and will not be read at a public opening. Written requests for proposal results must include the Request for Proposal number and closing date. If the bidder wishes to obtain a copy of the proposal tabulation and/or evaluation form(s) once award is complete, bidder should provide a self-addressed, stamped envelope with the proposal.

11.9 Proprietary Information

All evaluation documents for proposals are non-proprietary and subject to public disclosure after contract award. All proposal documents and information are subject to public disclosure under Ohio Revised Code Section 149.43. To exempt information provided in the proposal from public disclosure, bidders should identify any and all sections of their proposal they consider trade secrets or proprietary information. In the event of a public document request, the OHIO University Legal Affairs Office will review the sections so identified, and will make the final determination as to the need to disclose. Bidders will be solely responsible for protecting their own trade secret or proprietary information, and will be responsible for all costs associated with protecting this information from disclosure. The University will keep one (1) copy of proposals in accordance with its record retention schedule.

11.10 Use of OHIO University, Bowling Green State University or Shawnee State’s Name

No Supplier providing proposals, products or services to the Shared Services Project will appropriate or make use of any of the University’s names or other identifying marks or property in its advertising without prior written consent of each University.

11.11 Gratuities and gifts

Gratuities are not acceptable. OHIO University, by written notice to Bidder, may immediately reject any proposal, or cancel any contract that results from this RFP, if the University finds that gratuities were given or offered. Gratuities are defined as gifts, entertainment, or any other compensation offered or given by the Bidder, or any agent or representative of the Bidder, to an OHIO University, Bowling Green State University, or Shawnee State University officer or employee, in an effort to secure an award or preferential treatment.

11.12 Extension of Award to IUC-PG member school

Unless otherwise note, submission of your response indicated acceptance of the extension of the successful Bidder’s offerings and pricing to all members in the IUC-PG.


12 TIMELINE

August 23, 2011

Date RFP Distributed via IUC-PG OHIO University Sourcing Manager:

RFP distributed to vendors.

September 6, 2011 – 2:00pm EST

Vendor Questions Due:

Qualified vendors may submit question pertaining to the RFP to [email protected].

September 20, 2011

Responses to Vendor Questions:

OHIO University will distribute responses to vendor questions.

September 27, 2011 – 2:00pm EST

RFP BID RESPONSES DUE to

https://sciquest.ionwave.net/prod/vendorregistration.aspx?vcid=527) OHIO University will accept RFP responses.

October 3, 2011 – October 28, 2011

Response Evaluation and Scoring:

OHIO University will evaluate and score all responses to select top candidates.

October 31, 2011- November 30, 2011

Top Candidates Bake-Off:

Based on the scoring of the responses, the top vendors will be invited to give oral presentations

and product demonstrations for the selected solutions and to discuss any clarifications necessary

regarding their responses.

December 1, 2011

Top Candidate Selected:

OHIO University will announce the top solution for further evaluation, and will issue a letter of

intent indicating the desire to enter into a contractual agreement contingent upon a successful

pilot deployment.

December 1, 2011 – February 29, 2012

Project Pilot:

The top vendor will work with OHIO University OIT to develop and finalize the design. The

vendor will also be on-site to install and implement the necessary equipment and configurations

for the project pilot.

The pilot will be built using new equipment that will be sized and implemented as it would to

meet the full requirements of the project. This equipment will remain in service and will be

purchased by OHIO University if the pilot is successful.

mailto:[email protected]



If the pilot is deemed unsuccessful, all equipment will be returned to vendor and OHIO

University will bear no expense for the project; materials, labor, shipping or otherwise. The

University may examine other vendors or solutions.

March 1, 2012

Anticipated Award Date

2012

Contract start date dependent on executed contract

Documents

Attachment B - Netech Corp (Cisco) RFP Response - Inter-University