16720ip- Telephone System

7/27/2019 16720ip- Telephone System

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SECTION 16720

TELEPHONE SYSTEM

00000000PART 1 - GENERAL

1.1 DESCRIPTION:

A. The work shall consist of furnishing and installation of IP based telephone systemas shown on the drawings and specified herein.

B. Basic Criteria:

The concept design includes for the use of an IP based telephony system withanalog capability.

An IP telephony solution utilizes the building LAN and forms an integrated part of

an integrated IP network to deliver voice services as standard IP data streams. Themain advantages to the operations in employing this type of technology are theunified management of the voice service system (i.e. voice services can bemanaged and administered in the same way as the data network) and the inherentflexibility provided for moves, adds and changes.

It is also reasonable to consider that at some stage the TelecommunicationsRegulatory Authorities will permit external voice calls to be made via the internet.

In order to provide the greatest degree of flexibility the system will provide gatewaysincluding incoming and outbound calling, analogue ports for emergency and inhouse use, PSTN, ISDN to meeting rooms for videoconference use, Fax lines toAdmin and Business Centre and legacy TDM.

The Voice System shall support industry standards including H.323, 802.1p,802.1q, MGCP, TAPI, JTAPI and SIP. Of these possibly the most important issupport for the SIP interoperability standard as this will provide the flexibility of usinghandsets from any number of vendors supporting this standard.

The VoIP telephony system will provide the following minimum features:-

Attendant administration

Direct Outward Dialing

Direct Inward Dialing

Call Waiting

Call ID for incoming trunk and station

Direct trunk selection

Busy verification of station lines

Attendant control of trunk group access

Privacy line locks out

Attendant conference

Transfer and extension of calls

The attendant console keypad will provide tone signaling and alphanumeric foraccess to voice messaging and beeper systems

ITCC in RiyadhResidential ComplexJ10-13300

16720-1 Telephone System


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Message Box features. (Simplified Operation / Easy to use)

Music on Hold

XML / WML Customization

Voice over Wi-Fi shall be used to provide staff and customers with mobile

telephone serviceThe VoIP telephony system will support real time event management reportingincluding:-

Status trunking,

Status of all routing components,

Status of all remote components,

Status of all gateway ports

Calls trace capability.

An interface shall be provided to the PMS for Calling Party Name Display (CPND) to

be available on all phones in guest rooms so that the name, room number andrelated information of the calling parties can be displayed. The CPND informationwill be automatically updated over the interface from property Management System(PMS).

An interface shall be provided to the guest room management system to achievesome function via guest room bedside telephone to control HVAC, Lighting, etc. viaIP telephone set.

1.2 REFERENCES:

A. NFPA - National Fire Protection Association

NFPA 70 National Electric Code

NFPA 78 Lightning Protection Code

B. NEMA - National Electric Manufacturer Association

NEMA 250 Enclosures for Electrical Equipment

C. EIA - Electric Industries Association

RS-453 Dimensional, Mechanical and Electrical CharacteristicsDeferring Phone Plugs and Jacks

RS-464 Private Branch Exchange (PBX) Switching Equipment for Voiceban Applications

D. IEEE - Institute of Electrical and Electronic Engineers

Measuring Transmission Performance of Telephone Sets

E. UL - Underwriters Laboratories Inc.

All Relevant Standards




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F. CCITT - International Telegraph and Telephone Consultative Committee

All Relevant Publications

G. STC Saudi Telephone Company

Material Specifications

1.3 SUBMITTALS:

A. Product Data: Submit manufacturer's technical data on telephone systems andcomponents.

B. Shop Drawings: Submit layout drawings of telephone systems and accessoriesincluding, but not necessarily limited to, service-entrance penetrations, mountingpanels, apparatus, racks, terminals, and switching equipment.

C. Wiring Diagrams: Submit wiring diagrams for telephone systems, including rack andterminal connections. Also show wiring connections to electrical power feeders.

1.4 QUALITY ASSURANCE:

A. Manufacturer's Qualifications: Firms regularly engaged in manufacture telephonesystems and ancillary equipment, of types, ratings and capacities required, whoseproducts have been in satisfactory use in similar service for not less than 5 years.

B. Installer Qualifications: Specialist subcontractor with at least 5 years of successfulinstallation experience with projects utilizing telephone systems and equipmentsimilar to that required for this project. Subcontractor shall be subject to approval of

Engineer.

C. Materials and installation shall comply with the specified Codes and Standards.

D. Single Source Responsibility: All components and accessories shall be product ofsingle manufacturer.

1.5 TRANSPORTATION, HANDLING AND STORAGE:

A. Deliver telephone equipment and components in factory-fabricated containers orwrappings, which properly protect equipment from damage.

B. Store telephone equipment and components in original packaging. Store inside in awell-ventilated space protected from weather, moisture, soiling, humidity, andextreme temperatures.

C. Handle telephone equipment and components carefully to prevent damage,breaking, and scoring of finishes. Do not install damaged units or components;replace with new.

1.6 WARRANTY:

A. Submit manufacturers written warranty for period of 1 year from substantialcompletion for the repair or replacement of defective materials and workmanship.




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PART 2 - PRODUCTS

2.1 SYSTEM ARCHITECTURE

A. The architecture is unification of the voice and data communicationinfrastructures, to build-up a communication platform that is standards based,interconnecting the different elements of a distributed system. Elements such as:Control, Media Gateways with or without intelligence, IP phones, IP applicationphones, and servers; that work collectively as a single system over the samecommon transport infrastructure.

The communication platform required should provide the links necessary tointegrate multimedia and business applications via open standard interfaces thatdeal with the most recent IP and Web technology standards. Thetelecommunications system must be composed of two distinct parts: the softwarethat controls the communications and the hardware components supporting thecommunications interfaces.

B. Hardware: The system should offer great flexibility and support versatileconfigurations and types of Media Gateways. The same hardware componentsshould permit multiple architectural configurations:

PBX type centralized architecture of mixed architecture centralized /

distributed on network: or on network packet:

Switched) Circuit network

Copper cable Single mode or multi-mode fiber optic cable

Packet network

LAN / WAN IP network

In a distributed IP architecture, "Media-Gateways" must be configured withadequate resources (i.e. VoIP Codecs) to support the expected traffic coming fromthe configured non-IP end points in the system, including those for:

Digital stations

Analog stations

Wireless services: Radio base station for DECT type (if required)

Analog public network (PSTN) trunking

ISDN type and T2/T0 public network trunking (as required)

PBX-PBX automatic digital and/or analogue tie lines (as required)

Leased lines: complete or fractional T2 ISDN

To facilitate the integration of the systems in a structured plan, "Media-Gateways"can be packed as standard 19-inch stackable cabinets, and standard RJ45-typeconnections in order to allow consolidation of the voice and data elements via aunified wiring plan.

The CPUs shall either based on standard 19 industrial garde Servers (from IBM,




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HP Dell or approved equal manufacturers) or the standard Cards in MediaGateways. Both architectures shall support IP or TDM or mixed IP + TDMimplementations. Redendency (one duty and one standby) of servers and controlarchitecture shall be provided to ensure uninterruptible service. It is essential thattotal system support Carrier Class (99.99%) reliability is available.

C. Software: The proposed system shall be based on an open softwarearchitecture and standardized encouraging the functional integration thus withthe systems of information. Software control of the real-time communicationsshall be based on the LINUX operating system. This communicationsmanagement software or "soft-switch" should follow the standard"Appliance/server" platform,

D. Redundancy of Sensitive Elements: It is required that in addition to theinformation storage devices (hard disk), that the real-time switching software shallalso be duplicated (redundant). The architecture based on soft-switches shouldpermit the geographical relocation of the communication servers through astandard IP network. The interface cards should generate their own feeds from acommon source, and the analog and digital user should be individually equippedwith the supplemental devices necessary for proper DTMF operation, three-partyconference circuits, etc

The system shall offer maximum availability, with the switchover of a CPU or soft-switch to the other in case of a problem should conform with the model used incomputer systems: the complete "mirroring" of the information such as fixed orvariable data.

In addition to the configuration data and the dynamic data, (call status, call detailrecords, traffic collection, etc.) the complete set of programs and software modulesmust be duplicated in real time. In case of problems in the main system (hardware

or software), the standby system (emergency mirror) must take over the control ofcommunications instantaneously:

Within the same system, the switchover must be transparent to the users

The switchover must not initiate reloading of the system

At least internal and PSTN calls should not be cut

External calls on hold will be redistributed to the operators

The system will also be equipped with a device permitting the automatic backup(programmable) of the data and the programs necessary to be in good workingorder in a third system and which will be stored automatically in the Buyerorganization's computer back-up system.

E. Security: Dedicated technical areas are locked, controlling access to the systemand to the cabling system cabinets. The system must offer standard protectionfor password control features as well as system access by password / shadowpassword file / aging passwords. In addition to this basic rule, the system mustincorporate these other security aspects.

1. Controlled access to management platform: One of the most importantsecurity considerations is protecting access to system management. Thesystem must control the identity of the management terminals and the useraccessing that terminal. During a connection, (local or remote) the systemmust check the consistency between the management platform name,management platform password, and user name before authorizing theconnection.




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2. Secure remote access via a public switch exchange: The system must offerseveral secure management and maintenance access through the publicswitch exchange. The remote access proposed, either via PSTN or ISDN,must provide a high security level for remote management/maintenanceterminals at a predetermined location, as well as the normal user name andpassword control.

3. Remote access through Internet or Intranet: A firewall is the onlymechanism that can adequately protect the computing and communicationresources via Internet or Intranet access. But to protect and control remoteaccess through IP networks, the proposed system must support a filteringengine to control IP devices and services that have access to the system; alight and cost effective embedded protection mechanisms such as a trustedhost and TCP wrapper.

F. IP Communications

1. IP station support: The system must manage, control and support a rangeof IP telephone stations for both voice and telephony applications as wellas IP application stations for voice, telephony, and Web services support.IP telephone and IP application telephone stations will be referred herein asIP stations, and when necessary, the type of IP station will be clearlystated.

2. Quality of Service: The proposed system supports native IPcommunications in direct or "peer-to-peer" with only the telephone signalingtransiting back toward the controlling communications server. The speechwill be switched over the IP network and exchanged directly from client to

client. The voice and signaling frames should be marked [tagged] in orderto be recognized and should be classified by the network. The standards ofmarking supported will be:

Level 2: IEEE 802.1p /Q

Level 3: TOS / DiffServ

In the case of a PC connected to an IP station, (IP telephone or IP applicationtelephone), the frames transmitted by the PC, tagged or untagged, must betreated by the IP station in a transparent manner.

3. Client DHCP: The client IP Media Gateway (the IP stations) will support

either a static IP address or a dynamic address (manageable from theterminal) by the Clients compatible DHCP within their server.

4. Automatic VLAN Assignment: Although voice and data traffic flow overdifferent VLANs, they will be managed simultaneously because of theirdistribution over one network. When a user station is moved, its possible toperform the IP activation for a VLAN voice connection that is different fromthe one initially programmed in the terminal. In this case, thetelecommunications system should have the capability to support aprocedure, based on standard mechanisms, to assign the correspondingVLAN number to the IP station clients during IP station initialization.

5. IP Communication Zones: An IP communication zone (IP domain) isdefined by IP stations and / or media gateways within a geographical area.




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Because bandwidth is shared by the data and voice communications, itsnecessary to limit the bandwidth assigned. The goal is to take into accountthe underlying IP infrastructure to avoid congestion in the WAN links andmaintain good quality voice communications.

The system must be able to control and limit the number of simultaneouscommunications between the different communication zones. In addition, thecompression algorithms used for the communications inside the same IPcommunication zone and between different IP communication zones can bedifferent and must be activated on a call-by-call basis. For example, inside thesame IP communication zone (same LAN), communications will not becompressed but digitized in G.711 in order to have toll quality voice; whereasinter-zone communications will use the compression algorithm defined by theadministrator: G723.1 or G.729A.

6. System Availability: In the distributed IP architecture with a soft switchcontrolling the IP stations and the media gateways, their operation must bemaintained by an emergency arrangement. In case the IP network goesdown, the link with the soft switch must recover automatically via the analogpublic network or ISDN. In this case, the media gateway will behave like anautonomous PBX, and if some IP stations are located on the same site asthe media gateway, they will also recover their signaling to thecommunication server through the MG.

7. H323 Compatibility: The proposed telecommunications system mustsupport H.323 technology and should permit the following functions:

Management of communications between H.323 terminals

Interoperability between the H.323 terminal and the traditional telephony

devices (digital stations, IP, analog, private or public lines)

Gatekeeper Server: The proposed system should integrate an H.323gatekeeper server that offers the following services:

Automatic registration of the H.323 terminal and assignment of a call

number by the RAS protocol (registration admission status)

Resolution of the address, the terminal H.323 can be identified by its call

number or by its IP address that can be assigned dynamically by a DHCPserver

Establishment of communications in direct mode

If it is necessary to communicate with another external gatekeeper (Internet,LEC/CLEC/IXC, corporate, etc.), the system should be capable of registeringitself with these entities.

H.323 Gateway: The proposed system should include a gateway thatallows the Buyers H.323 devices to interoperate with the traditionaltelephony devices (digital stations, IP, analog, private or public lines) andthe SIP terminal. The main H.323 gateway features required are:

Support of the H.225 and H.245 protocols

"Fast-connect" setting

H.245 tunneling Registration, admission, status (RAS)




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Direct mode and routing mode

Connection to an external gatekeeper

H.323 attachment D for T38 fax

Direct RTP

8. SIP compatibility: The proposed system should permit the integration of SIPterminals with other terminals and private or public external lines used bythe company. The SIP software should conform to the normalizedarchitecture and be integrated in real-time communications management tobenefit from the duplicated services. The SIP modules are:

SIP Proxy

SIP Registrar

SIP Gateway

The SIP terminal will use either UDP or TCP to communicate. The supportedstandards must conform to the following RFC:

RFC 3261 SIP: Session Initiation Protocol

RFC 3262 Reliability of Provisional Responses

RFC 3264 An Offer/Answer Model with the Session Description Protocol

(SDP)

RFC 3265 SIP Specific Event Notification

As well as the RFC in development (draft) such as:

Transfer:

o draft-ietf-sip-refer-06.txt

o draft-ietf-sip-replace-02.txto draft-ietf-sip-cc-transfer-05.txt

Message on hold: draft-ietf-sipping-mwi-01.txt

Draft-ietf-sip-session-timer-09

RFC 2833: useful RTP data for DTMF numbers

9. SIP terminal support: The proposed system should allow the SIP terminal toregister itself on the system via the SIP proxy module AND to see itself. Itsalso assigned a directory number in the communication server so thattraditional telephone services are also available. The following services arerequired to be available to the SIP terminal:

Caller name and number of the caller or the requestor (on the terminal

screen)

Hold

Retrieve from hold

Transfer

Unconditional forwarding

Conditional forwarding on busy or no-answer

Three-party conference

Do not disturb

DTMF signaling (in band or out-of-band, the RFC 2833 protocol)

Message waiting indication for the voice messaging system

Classes of service for external calling




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Call records for external calls

10. Authentication: An HTTP Digest (MD5) authentication procedure must bedefined between the terminal and the SIP proxy of the Buyer organization,or any other SIP point, or external proxy SIP, at the time the call or

message is initiated or in operation.

11. External SIP communications: The SIP terminals in the organization, aswell as the traditional stations should be able to communicate with facilitiesbehind a SIP connection or external SIP proxy (service provider class orenterprise). All SIP calls must pass through the system SIP proxy, whichcontrols the maximum number of calls. It must also be possible to arrangeinterception and prohibition of calls from unknown SIP terminals to all Clientstations.

G. XML/VxML Compatibility: The proposed telecommunication system should allow

the use of high level XML APIs based on Web technology standards

(XML/SOAP/ VxML) to ease creation of telephony and call control features forintegrating telephony services into web applications. The solution must be able tohandle a high capacity of clients using XML services. The access to the XMLtelephony application must be protected by user login name and password. The

following XML telephony features are required to be available to users equippedwith one of the following phone types: analog, digital, wireless (DECT/PWT), andIP telephone stations.

Login / logoutPasswordTelephony services

Multi-line Make call

Take call

Clear call

Transfer

Conference call

MF sending

Call progress information

Forwarding

Immediate No answer

Busy

Busy or no answer

The IP application station that accesses system telephony services and webservices from corporate or external Web application servers must also be XMLcompatible.

H. Connectivity interfaces: The telecommunication system will support the followingconnectivity:

Public Switched Telephone Networks (PSTN):




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Analogue Trunks (Ground start, Loop start, etc.) with polarity reversal /

Metering Pulses

Analogue Trunks with Caller ID

Analogue DID code R2

Digital DID code R2

ISDN Basic rate access ISDN Primary rate access T2

Frame relay interface

Ethernet TCP-IP interface

Pubic Packet Networks

X24/V11 or V36 Frame relay

Ethernet 10 BT

WAN Voice

Analogue Tie lines, 2,4,6 wires, E&M, 50HZ, L1, DC5A protocols

Digital Tie lines, E&M, 50HZ, L1, DC5A protocols

ISDN Basic rate access, private network protocol, QSIG

ISDN Primary rate access, private network, DPNSS protocols

ISDN Fractional primary rate access, private network, QSIG, DPNSS

protocols

WAN Data

ATM, 2 MBPS/VTOA, private network protocol

X24/V11 or V36 Frame relay, Private network, QSIG protocols

Ethernet TCP-IP, Private network protocol

Terminal connectivity

Analogue interface for analogue stations and fax machines

Digital interface, for proprietary stations

ISDN 2B + D, interface bus for S0 stations

IP telephone stations

IP application stations

H.323 terminals

SIP terminals

2.2 SYSTEM SIZING

A. The system shall capable to support all stations shown on drawings with 25%spare capacity for future growth. The system shall be expandable for adding newstations in future.

2.3 SYSTEM FEATURES

A. Unified Communications: The system will offer a complete suite oftelecommunications applications that enable users to control and manage their

calls, voice messages, email, fax, directories, and collaboration tools. Unifiedaccess to these applications will be possible via an HTML interface, or through




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the use of a telephone terminal.

B. The system shall include the following features:

1. Last number redialing2. Company directory3. Music on hold4. Hands free telephone sets5. Paging interface6. Manual Message waiting7. Display the dialing number and intercom calling station number for some sets8. Ring again9. Follow me10. Facilities of tie lines with other PABX11. Facility of using extension lines as data lines for computer12. Classification of extensions13. Extension to operator calls14. Outgoing calls15. Booking outgoing calls16. Operator to extension calls17. Inquiry calls18. Transfer calls19. Conference calls up to 8 parties per call20. Priority21. Common night service22. Universal night service23. Individual night service24. Direct in lines25. Automatic call back

26. Individual call diversion27. Individual call diversion when no answer28. Flexible numbering29. Line lockout30. Choice of individual trunk line31. Group hunting32. Control of long-distance dialing33. Direct speech connection34. Call pick up35. Grouping of trunk lines36. Call waiting indication to extension37. Common abbreviated dialing

38. Hot line39. Trunk to trunk transfer40. Calling names numbers display on digital instruments41. Incoming external calls with:

Camp-on busyBreak-in and forced releaseOperator recall

42. Additional Facilities:

Push-button dialing

Night time classification of extension




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2.4 ATTENDANT (OPERATOR) POSITIONS:

A. Propose a range of attendant stations for Engineers approval, with a precisedescription of the features provided by each of them. Attendant terminals arerequired to have a color screen, as well as attendant station software that can beused on PCs that are not dedicated to the attendant operation.

B. The answering device will be designed specifically for answering calls; thepresentation of calls; chaining of calls, and re-dialing will be entirely controlled bythe communications system. Nevertheless, it will be possible to hold specificindividual calls on keys reserved for this purpose. The capacity of the differentwaiting queues should be unlimited. Signaling or indications displayed on theterminal should be sufficiently explicit and unambiguous to facilitate the handlingof calls, and will give a maximum of information about the calls (normal, urgent,status of the waiting queues, name of internal callers making requests, status ofthe terminal, etc.)

C. Propose a device that allows the operators of the attendant positions tocontinuously supervise (view) the state of some stations in the installation (free,busy internal, busy external, unavailable). There must be at least 40 supervisedterminals. The attendant will be able to modify the mode of answering calls attheir answering console. In automatic answer mode, calls coming into theattendant console will be presented and connected without requiring manualintervention by the operator

D. The specific central answering position requirements are specified below:.

Call by name to internal or external parties

Text messaging

Hands-free and amplified listening Ringer levels

DTMF Signaling

Attendant withdrawal from attendant group

Withdrawal of last attendant

Multiple attendant positions

Automatic switchover of the attendant consoles

Attendant position locking

Call recording

Directories access

Management services

2.5 TELEPHONE TERMINALS (HANDSETS)

A. General: Submit for Engineers approval a range of digital telephone equipmentas well as IP stations, each capable of meeting one or more of a variety ofneeds, ranging from basic equipment for use in public areas to that of businessor management class. Type and number of telephone terminals shall be asshown on the drawings or as approved by Engineer.

B. Analog Terminals: The analog terminals shall be able to respond to either pulseor DTMF dialing signals, and automatically recognize the type of signal being

received.

C. Analog Terminals with Caller ID: Any standard analog Call ID telephone from well




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known manufacturer suitable for interfacing with the system. The terminal shallshow the incoming internal and external caller ID in case CLI information isavailable on analog/digital/ISDN trunks. The analog terminals shall be able torespond to either pulse or DTMF dialing signals, and automatically recognize thetype of signal being received.

D. Digital Telephone Stations: Propose a range of digital terminals for Engineersapproval and provide precise descriptions of the features available with each ofthem. Only electronic digital terminals will be accepted. For each type of terminalproposed, specify if the terminals support the following features:

Power feed over the twisted pair wiring

Maximum operating distance on 22 and 24-guage twisted pair wiring

Distribution on a single pair

Communications at 256 Kbps between the terminal and the equipment

An additional analog interface

The additional analog interface must provide ringing generator, power feeding, inthe following configurations:

Common or different extension number

Analogue interface dial independent of the host terminal

Analogue interface dial for the host terminal

E. IP Telephone Stations: Propose a range of IP telephone stations for Engineersapproval. These IP telephone stations shall be identical to the digital stations inaspect and function. For every type of station proposed, specify if the stationssupport the following features:

Remote power feed per the 802.3af standard or local 120 / 230 -volt feed

Auto-sensing 10/100 Ethernet switch interfaces

2 Port 10/100 switch

QoS (Internal the station and priority to the voice signal

Frame marking voice level 2 802.3 p / Q and level 3 ToS / DiffServ

Transparent recovery of frames by the associated PC (not by the station)

Fixed or dynamic assignment of the IP address by customer DHCP

Agent SNMP integrated with MIB 2

Voice compression standard G711, G723.1, G729a

1 Type 1: Executive Telephone(Supervisory or user positions handling heavy traffic)

Executive IP Phone supports a rich suite of endpoint applications, including XML-based applications on all models.

Selected models also support MIDlet-enabled applications, UnifiedCommunications Widgets, applications developed within the Unified ApplicationEnvironment and personal video communications.

High-definition (HD) voice, vibrant color displays, Gigabit Ethernetconnectivity, and more than basic support for endpoint applications.

Executive IP Phone supports multiple-call per-line appearance, users can takeadvantage of advanced call navigation capabilities with support of multiple-call




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sessions on a per-line basis. For example, on a 2-line endpoint, you can be onan active call and navigate to pick up a second incoming call on the same line,while the first call is automatically placed on hold then user can switch back andforth between these two call sessions, as required.Executive IP Phone supports a rich suite of endpoint applications, including XML-based applications on all models.Selected models also support MIDlet-enabled applications, unifiedcommunications widgets applications developed within the unified applicationenvironment and personal video communications.

Features:- Eight programmable backlit line or feature- Four-way navigation cluster features- keys deliver quick access to communications- A large 5.6-inch, high-resolution, 320 x 240 pixel graphical color

display with touch screen for superior features and applicationdetail and interaction

- High-definition voice (HD voice) support of headset, handset, andfull-duplex speakerphone for superior audio performance

- Integrated IEEE 10/100 switch ports support co-location of a PC atthe workspace

- Contrast Control- XLM Applications- HD Voice- Unified communications- Hands-free and amplified listening modes- LCD icons for signaling associated with every key- Volume control of the handset receiver- Integrated Help Service to aid user in programming

- Access to ISDN services- Model: CISCO IP Phone 7975G or any operator approved equal

2 Type 2: Intermediate Telephone(Stations for assistants or users handling medium traffic)

The intermediate IP Phone is an ideal endpoint for users who handle a largevolume of voice calls. Which well be suited for Administrative staff,managers, contact center agent and supervisor.

Feature:

- Easy viewing under varied lighting conditions:- Backlit, antiglare, pixel-based graphical- Monochrome LCD displays optimize readability.- Convenient, hands-free communications:- Full-duplex speakerphones and dedicated headsets give users

more control and more communications options.- At-a-glance call status indication: Tricolor, illuminated line keys

provide quick call state recognition.- Enhanced user experience: Rounded, ergonomic keys deliver a

superior tactile feel, and choice of headset styles delivers greaterconvenience and comfort.

- Fixed keys: The endpoints have fixed keys for fast access to

commonly used functions such as Directory, Settings, Transfer,Conference,




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- Hold and Messages.- Single-call per-line appearance: These endpoints offer traditional

telephony call interactive experiences.- Integrated 10/100 Ethernet switch: The endpoint switch supports

co-location of a PC at the workspace.- Localized language orientation: Right-to-left language

presentation provides support for global deployments.- Guided interaction: Four dynamic soft-label keys guide users

through call features and functions.- Healthcare organizations- Financial institutions- Hospitality businesses- Twelve tricolor illuminated line keys for quick- Call status identification- Customizable, paper-label insert for one touch- Access to commonly used features- Easy-to-read, 396 x 81 pixel, white-backlit- Monochrome antiglare LCD display optimizes- Viewing under a variety of lighting conditions- Model: CISCO IP Phone 6961 or any operator approved equal

3 Type 3: Basic Telephone(Stations for users with average traffic)

Basic telephone is a single-line Unified SIP phone an affordable,entry-level phone that is well-suited for settings with low tomoderate voice communications usage.

Features:

- Half-duplex speakerphone and internal microphone for clearcommunications

- Fixed feature keys for one-touch access to redial, transfer,conference, hold, line select, mute, speakerphone, and voicemail

- LED's provide quick call status indication at-a-glance- Supports a 2-line by 24-character monochrome display- Two menu select keys and a multiparty rocker for scrolling control- Supports unified communications- Additional capabilities such as caller ID, call history and the ability

to configure the phone- Choice of Power over Ethernet (PoE) or local power through an

optional power adaptor

- Model: CISCO SIP Phone 3911 or any operator approved equal

F. IP application Stations: Submit for Engineers approval, a range of IP applicationstations full-featured with integrated IP connectivity and telephony, capable ofsupporting any web-based business application that is compatible with XML. Atleast 2 IP application stations shall be provided, unless indicated otherwise Foreach type of terminal proposed, provide precise descriptions of the featuresavailable with each of them as well as specify if the terminals support thefollowing features:

10BT/100BT connection: half/full duplex with auto negotiation

PC port

VoIP standard: H.323 voice compliance, RTP, RTCP

Voice compression standards: G711, G723.1, G729a




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Integrated Ethernet switch with QOS support Frame Marking voice level 2

802.3 p / Q and level 3 ToS / DiffServ

Support for both: External power supply and Ethernet cabling system IEEE

802.3af full standard compliant

Fixed or dynamic assignment of the IP address by customer DHCP

IP application stations should be equipped with:

Large color or black and white adjustable screen

Context-sensitive keys associated to the display These contextual keys are

linked to the context displayed on the screen to directly activate functions

Navigation keys to navigate inside the graphical interface (change of

application or context, return to the home page)

Numeric keypad to dial a number or enter digits in a entry field

Integrated alphabetic keyboard for functions such as text messaging and dial

by name

Full duplex hands-free mode with echo cancellation Audio operation to tune audio levels, mute, loudspeaker,

Connector for headset or additional speakerphone

Automatic and transparent switch from one to another communication mode

(headset, handset, hands-free, etc.)

Wireless Bluetooth capabilities (Based on 1.1 Bluetooth specification)

Open to applications: Access to corporate or external Web based application

via third party SDKs, APIs (XML, )

Options:

10 or 40-key add-on modules

External conferencing device (with Bluetooth or cable)

IP application station shall be an open platform allowing the integration of corporate,external, hosted or third party Web applications via XML/SOAP. It shall provide aset of tools customizing their communications to the specific day-to-day demands ofwork and to fit specific enterprise, group and individual needs.

2.6 POWER SUPPLY

A. Power Supply: Device for controlling the designated minimum and maximumvalues of the continuous power delivered. When one of these thresholds isreached, the energy feed to the telecommunications system must be cutautomatically. Control devices will provide both automatic and manual control of

the float and balancing, with automatic return to float mode when the battery hasbeen charged to the required level. The rectifier and power supply will be sized tofeed the power requirements of the telecommunications system at thedesignated cabled capacity, and should be able to recharge the batteries within10 hours

B. Battery: Batteries in the power supply will be the maintenance-free type. Batterieswill be installed to power the telecommunications system automatically in case ofa failure of the electric power or the rectifier. This uninterruptible power supply willmaintain the telecommunications system at full traffic for a minimum period offour hours.

C. Distributed Power Architecture: For an IP network with a distributed architecturewhere the media gateways power is connected directly on the sector, an UPS




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should be integrated to provide standby operation of at least for two hours.

2.7 SYSTEM ADMINISTRATION:

A. Management Terminal: The system management server will be based on thelatest technologies, such as JAVA/JEE. Provided at least one managementserver, unless indicated otherwise. The server shall support a minimum of five (5)clients having different access rights to the applications. Access to the servermust also be possible by a light Netscape or Microsoft Explorer client. Themanagement server will offer a suite of integrated management tools:

System programming

Control and allocation of communications costs

Traffic data collection

Alarm system synchronized with the topology (For muti-site application only)

Topology with auto-discovery

Company directory

The set of these applications must be interactive. For example, any modificationto the telephone data will automatically generate a change in the telephone data,and vice versa, any modification of the user station parameters will generate anupdate in the directory.

For example, all directory data that is modified will automatically update thetelephone system and reciprocally, all subscriber parameter changes will updatethe directory. A user shall also have a complete station information in a window:

Management parameters

Information on communications costs Information on telephone traffic

B. Connectivity and Modem Access Security Management: The management servermust be able to be situated anywhere on the premises, and accessible to thecommunications system(s) via an IP network, the PSTN, or ISDN. Thetelecommunications system must be able to control remote access coming in viathe PSTN or ISDN. In an ISDN (CLIP) environment, the system must be capableof authenticating the identification of the requestor transported by the network. Ifthe number does not correspond to an authorized access, the request is deniedand an alarm is generated, indicating the calling number information for theunauthorized request. In the PSTN environment, when the network does not

transport information, the remote (CLIP) must present a user name and passwordfor the connection. If the identification is correct, the system releases the call andcalls a predefined number in memory that corresponds to the user name.

C. Management Access Security: The access to the services offered by themanagement server should be protected in the following manner:

Authentication of the management server

Authentication of the client user

Authentication of the server and customer should be based on a user name andpassword. During the connection phase, the voice communications system

should test the coherency of three items:




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Name of the management server

Management server password

User name

To ensure connection and information exchanges securely, the management

server should employ sftp and scp SSH/SSL for telnet sessions. In addition, itshould use IPsec mechanisms between the server and the management clients.The management server authenticates the user password in order to permitdomain management.

D. Domain Management: A primary administrator who has access to the suite ofapplications and information residing in the management server (includingmasked numbers and personal codes) can define domain management. Thisprimary administrator will have the right to create user profiles defining thefollowing:

Access rights to one or more applications: yes or no (e.g., telephone cost

administration) Access level to the applications: read, read / write, or read / write / create /

delete

E. System Management: The telecommunications system configuration andmanagement application must permit management of all the parameters via agraphical user interface. The configuration application should have the followingfeatures:

Independence in relation to software releases for the communication system,

but with automatic synchronization with the new releases.

Capability to configure several systems in the same session

The configuration will be able to operate in a batch mode via import / export

utilities

Selection of items with the help of menus and automatic modification of all

the elements or of selected items

Configuration of the parameters of an item (e.g., user) from a single window

Configuration of digital, and IP stations via a graphical interface

Configuration of the messaging system directly from the application

List of management actions giving the date and list of operations performed

F. Traffic History and Collection: The system administrator shall have access to dataregarding system traffic, use of telephone features, and overall system operationat all times. Collection of traffic data is required for measuring system use and todetect possible system sizing problems. The application results must bepresented in a graphical format. Bidder will describe and propose the varioustraffic history approaches possible. The specific needs are as follows:

Collection of the following elements by hour, hour, day and per month:

Speed of answer by the called stations

System station traffic

"Top ten" users by cost or by call length

Attendant group traffic

Holding time waiting for attendant(s)

External trunk traffic bundled in trunk groups

Traffic on individual trunks




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Data collection on the components that ensure the telecommunication

system's proper operation

Data collection on client IP traffic (station or media gateway) or between two

IP addresses defined by the administrator seeking the following information:

Number of frames exchanged Number of lost frames

Jitter

Delay

The traffic history data will be protected, by period, on a storage device thatpermits the reloading and subsequent use of the information.

G. Communications Cost Management: Call record storage on the system hard diskshall be for a minimum of 100,000 call records. Management and sorting of thesetickets will be performed using the management and traffic analysis applicationresiding on an external server. Call records will be automatically downloaded by

the application at night; for security reasons, a sliding file of the most recentseven (7) days will be maintained continuously on the system hard disk.

H. General Characteristics: On ISDN connections, the proposed system will not usedirect or per-call costing means from the carrier. On calls transferred by a stationor attendant, outgoing call costs should be assigned in a manner that accuratelyreflects the amount of time each party was in control of the call. For reasons oforganizational confidentiality, some stations will never be tracked for any reason;the costs generated by these stations will, nevertheless, be reflected on the costcounters. If it is not possible to receive cost reference information from thecarrier, the call accounting management system must be able to accuratelycalculate communications expenses by an alternate means. The application mustbe able to handle up to five carriers (service providers) simultaneously; asimulation utility integrated into the application will permit the system to use thestored information to identify the lowest cost service provider for external traffic.Indicate the various types of information storage and will specify the followingelements:

The type of support and the maximum capacity of record storage capacity

Storage devices

The different alarms that inform the attendant(s) of heavy memory use. Two

successive thresholds are required: 75% and 90% of the storage capacity.

To facilitate the overall management, automatic updating of the application isrequired to reflect when a user's parameters (e.g., number, assignment, deletion,etc.) are changed in the telecommunications system.

I. Organization: To adapt to the Client organization, the application should permitmultiple organizations with up to eight levels each. Access to each of the levels inan entity will be granted to multiple users as needed (department director,general manager, etc.) by the system administrator. These users will be assignedan access code that provides access to all the system features (such as creatingreports, program a prerecorded observation, etc.) for their organizationalgrouping. The ability to manage as many as 10 users, in addition to the primaryadministrator, shall be part of the feature. The hierarchy information must be

automatically synchronized when the telecommunications system undergoes arelevant change (stations, attendant(s), trunking, cost centers, etc.) to reflect the




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organizational structure. The hierarchy information must be printable.

Call detail records for cost reports: For every outgoing call, the system shallprovide the following information:

Caller name

Name of the service

Department name

Number dialed (with masking of the last four digits)

Date, hour, and minute that the call ended

Length of the call

Number of cost units

Cost for the call

Type of call

Service features used

For every incoming call, the system should provide the following information:

Name of called station

Name of the service

Department name

Caller number when it exists

Date, hour and minute that the call ended

Length of the call

Type of call

ISDN service features used

The proposed service shall offer the attendants a set of ready-to-use pre-definedreports, in addition to a management report utility to customize the existingreports or to create completely new ones. It will be possible to create graphs fromthese reports, in pie and histogram format. The following points will bedeterminants in the selection of the call cost accounting system:

Invoicing utility with data import and (where needed) removal of taxes or

other designated charges

Cost utility for allocating stationary costs of different items

Alarm reporting service, to indicate when defined parameters are met or

thresholds are exceeded on the following factors:

- Call numbers, costs, or length, or important variation in these data- Length of time a call remained on hold, abandoned DID calls- Volume of IP packets transmitted and received, and abnormal packets

When the various thresholds are reached, alarms must be generated, and emailadvice should be sent to one or more administrators.

J. Management of the System Alarms: The management server should offer anapplication that centralizes the alarms and relevant systems communicationevents, as well as any generated by the management server itself. These events

must be filtered and displayed in real time, according to the administrator'sneeds.




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These alarms should be categorized according to the six-level ISO severitydefinitions, and identified by different colors to facilitate direct attention to specificcategories. Every alarm should have a detailed explanation at the second level;in addition to likely causes, the application should indicate the appropriatetroubleshooting actions to resolve the problem.

In the case of a major alarm, an email must be automatically sent to one or moresystem administrators. The list of alarms and events, as well as their detailedexplanations must be printable and archive-able. A report generator will providestatistical reports.

K. Topology (For multi-site application only): The management server shouldprovide an application that offers a topological view of the telecommunicationssystem that constitutes the organization's network, as well as the links that existbetween sites. The presentation should be the simplest possible, and considerthe need for an automatic discovery utility that will automatically display the sub-networks and nodes that exist. This application must be completely customizablein terms of the screens and the icons that represent the objects. The differentobjects that represent the organization's network must be synchronized with thealarms application in real time, and items that are affected by alarm-generatingconditions must indicate the event, using the same severity and color format.

When a problem appears, the administrator will access the faulty element by amouse-click, and will move through the system architecture, following the networkhierarchy (node, cabinet, media-gateway, board, etc.) until the problem elementis found. The administrator will be able to automatically launch systemconfiguration and maintenance utilities while viewing this element.

Printing of reports: The management application should allow the automatic

printing of all types of reports on a daily basis, in addition to being able to send itby email in different formats, including:

Text format: .txt

PDF format: .pdf

HTML Format

Excel format: xls

2.8 VOICE MESSAGING

A. The basic telecommunications system must be equipped with a voice messagingapplication. An integrated system is preferred because it provides the bestpossible inter-operation. Describe all the services available to the users and thevoice messaging system administrator, the maximum number of voice mailboxes,the maximum recording capacity, and number simultaneous links that can behandled. In addition to the required services described below:

1. Answering or answering with date stamp: The system shall provide voicemailbox holders the choice of two functions: answering the messages oranswering them with a date stamp.

2. Customizable Announcements: When a call is forwarded to the voicemessaging system, the box holder will be able to choose between twopersonalized announcements. If the personal announcement has not been




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recorded, the standard system announcement will be substitutedautomatically.

3. Call forward to the voice messaging system from a box holder: When a boxholder agrees to have his or her call forwarded , or after a set amount oftime that there is no response, the internal or external caller is automaticallydirected to the voice messaging system. After the caller hears theannouncement, the caller can record a message or decide to be forwardedto an attendant or a specific station.

4. Overflow of Internal Calls: After a set amount of time, the system will offervoicemail to internal callers for those who have a voice mailbox.

5. Recording of Calls: The holder of a voice mailbox will be able to takeadvantage of this service to record internal or external calls. Recorded callswill receive the same service as messages that have been left by callers.The minimum recording length will be 10 minutes; This service shall beautomatically provided to digital station users.

6. Message waiting Indication: The box holder is advised, by a lit lamp on thestation (either digital or analog equipped with a message lamp) that thereare messages in the mailbox. Users who have terminals without a messagelamp should be advised by a voice prompt that there are messages waiting,when user picks up the handset to make a call.

7. Message Review or Access: The box holder can review or access waitingmessages from any internal station or an external station via the generaltelephone network. When using a digital terminal, the display will offerdynamic context-sensitive soft keys that operate the message system

features on an interactive basis: fast forward, stop, rewind, storage,continue, previous message, etc.

8. Access security: Security and privacy for the recording of personalannouncements and in the review of messages will be assured by apersonal code.

9. Spoken Newspaper: A spoken word "newspaper" or "information" servicewith a minimum length of 30 minutes should be available.

10. Broadcast Lists: The box holder will have access to general and personaldistribution or broadcast lists.

11. Forwarding of Voice Mail Messages: The box holder will be able to send acopy of previously received messages to other boxes (with or withoutrequesting acknowledgement of receipt).

12. Message Storage: The messaging system will store messagesautomatically, for a programmable period of time, after which the messagewill be automatically erased; only a specific action to erase a message willoverride the storage of the message, and erase it immediately.

13. Call by Name: To provide universal access, it must be possible to select avoice mailbox by its name by using the telephone dialing keypad. The caller

will be guided in this operation by voice prompts. The service should be asavailable to an outside caller, in which case, the caller will be given the




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choice to deposit a message or having the system call the desired partydirectly.

14. Multiple Languages: To ensure consistency with the system voice prompts,the proposed system should be multi-lingual, offering four differentlanguages (as selected by Client). The service voice prompts played to auser will follow the same language the user has selected in thetelecommunications system. In no case will a mix or combination oflanguages on the same voice prompt be accepted.

15. Automated Attendant: The messaging system should offer an integratedand interactive automated attendant service. The application of messagingshould propose a service of integrated and interactive automatic operator.The user either calling or routed to the AA will receive a welcome message,and be able to select a language for the following interactions. Two (2)different languages must be supported by the application.

2.9 RACEWAYS , WIRES AND CABLES:

A. For raceways/trunking refer to Section 16050 - BASIC MATERIALS AND METHOD.

B. Other Wires and Cables: Refer to Section 16715 VOICE AND DATACOMMUNICATION CABLING.

PART 3 - EXECUTION

3.1 INSTALLATION:

A. The entire system shall be installed by specialist subcontractor approved by theEngineer.

B. Installation shall be in accordance with the approved drawings and manufacturer'swritten instructions.

C. The incoming external cables will be jointed after entering the building with dryplastic type cable prior to connection to the MDF (Main Distribution Frame).

D. For excavation and backfilling refer to Section 02200 - EARTHWORK.

3.2 ELECTRICAL SAFETY:

A. Separation of telecommunication circuits from the building electrical system andelectrical equipment shall conform to the latest publications of Articles 800-3 (a) and820-13 of National Electric Code (NEC).

B. An earth or ground shall be provided and extended to the termination box andconnected to each station protector. The earth or ground shall be installed andbonded in accordance with Article 250, 800 and 820 of National Electric Code(NEC) and conforming to Section 16452 - GROUNDING..

3.3 TESTING:

A. After installation of entire system and prior to acceptance of work, manufacturer'sstandard tests shall be conducted in the presence of the Engineer to show proper




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operation of each equipment, and the system entirely.

3.4 CONNECTION TO SAUDI-TELEPHONE LINES:

A. The contractor shall be responsible for coordinating and follow-up with SaudiTelephone Company for telephone connections and provide them requiredinformation and help to complete all paperwork. Charges for telephone connection(by Saudi Telephone Company) shall be borne by the Owner.

END OF SECTION

ITCC in Riyadh

16720ip- Telephone System

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Transcript of 16720ip- Telephone System