SECTION 16720 TELEPHONE SYSTEM 00000000PART 1 - GENERAL 1.1 DESCRIPTION: A. B.

The work shall consist of furnishing and installation of IP based telephone system as shown on the drawings and specified herein. Basic Criteria: The concept design includes for the use of an IP based telephony system with analog 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. The main advantages to the operations in employing this type of technology are the unified management of the voice service system (i.e. voice services can be managed and administered in the same way as the data network) and the inherent flexibility provided for moves, adds and changes. It is also reasonable to consider that at some stage the Telecommunications Regulatory 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 gateways including incoming and outbound calling, analogue ports for emergency and in house use, PSTN, ISDN to meeting rooms for videoconference use, Fax lines to Admin 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 is support for the SIP interoperability standard as this will provide the flexibility of using handsets 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 for access to voice messaging and beeper systems
ITCC in Riyadh Residential Complex J10-13300 16720-1 Telephone System

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 service The VoIP telephony system will support real time event management reporting including: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 and related information of the calling parties can be displayed. The CPND information will be automatically updated over the interface from property Management System (PMS). An interface shall be provided to the guest room management system to achieve some function via guest room bedside telephone to control HVAC, Lighting, etc. via IP telephone set. 1.2 REFERENCES: A. NFPA - National Fire Protection Association NFPA 70 NFPA 78 B. National Electric Code Lightning Protection Code

NEMA - National Electric Manufacturer Association NEMA 250 Enclosures for Electrical Equipment

C.

EIA - Electric Industries Association RS-453 RS-464 Dimensional, Mechanical and Deferring Phone Plugs and Jacks Electrical Characteristics

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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Telephone System

F.

CCITT - International Telegraph and Telephone Consultative Committee All Relevant Publications

G.

STC Saudi Telephone Company Material Specifications

1.3

SUBMITTALS: A. B. Product Data: Submit manufacturer's technical data on telephone systems and components. Shop Drawings: Submit layout drawings of telephone systems and accessories including, but not necessarily limited to, service-entrance penetrations, mounting panels, apparatus, racks, terminals, and switching equipment. Wiring Diagrams: Submit wiring diagrams for telephone systems, including rack and terminal connections. Also show wiring connections to electrical power feeders.

C. 1.4

QUALITY ASSURANCE: A. Manufacturer's Qualifications: Firms regularly engaged in manufacture telephone systems and ancillary equipment, of types, ratings and capacities required, whose products have been in satisfactory use in similar service for not less than 5 years. Installer Qualifications: Specialist subcontractor with at least 5 years of successful installation experience with projects utilizing telephone systems and equipment similar to that required for this project. Subcontractor shall be subject to approval of Engineer. Materials and installation shall comply with the specified Codes and Standards. Single Source Responsibility: All components and accessories shall be product of single manufacturer.

B.

C. D. 1.5

TRANSPORTATION, HANDLING AND STORAGE: A. B. Deliver telephone equipment and components in factory-fabricated containers or wrappings, which properly protect equipment from damage. Store telephone equipment and components in original packaging. Store inside in a well-ventilated space protected from weather, moisture, soiling, humidity, and extreme temperatures. Handle telephone equipment and components carefully to prevent damage, breaking, and scoring of finishes. Do not install damaged units or components; replace with new.

C.

1.6

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

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Telephone System

PART 2 - PRODUCTS 2.1 SYSTEM ARCHITECTURE A. The architecture is unification of the voice and data communication infrastructures, 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 application phones, and servers; that work collectively as a single system over the same common transport infrastructure. The communication platform required should provide the links necessary to integrate multimedia and business applications via open standard interfaces that deal with the most recent IP and Web technology standards. The telecommunications system must be composed of two distinct parts: the software that controls the communications and the hardware components supporting the communications interfaces. B. Hardware: The system should offer great flexibility and support versatile configurations and types of Media Gateways. The same hardware components should 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 with adequate resources (i.e. VoIP Codecs) to support the expected traffic coming from the 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-type connections in order to allow consolidation of the voice and data elements via a unified wiring plan. The CPUs shall either based on standard 19 industrial garde Servers (from IBM,
ITCC in Riyadh Residential Complex J10-13300 16720-4 Telephone System

HP Dell or approved equal manufacturers) or the standard Cards in Media Gateways. Both architectures shall support IP or TDM or mixed IP + TDM implementations. Redendency (one duty and one standby) of servers and control architecture shall be provided to ensure uninterruptible service. It is essential that total system support Carrier Class (99.99%) reliability is available. C. Software: The proposed system shall be based on an open software architecture and standardized encouraging the functional integration thus with the systems of information. Software control of the real-time communications shall be based on the LINUX operating system. This communications management software or "soft-switch" should follow the standard "Appliance/server" platform, Redundancy of Sensitive Elements: It is required that in addition to the information storage devices (hard disk), that the real-time switching software shall also be duplicated (redundant). The architecture based on soft-switches should permit the geographical relocation of the communication servers through a standard IP network. The interface cards should generate their own feeds from a common source, and the analog and digital user should be individually equipped with the supplemental devices necessary for proper DTMF operation, three-party conference circuits, etc The system shall offer maximum availability, with the switchover of a CPU or softswitch to the other in case of a problem should conform with the model used in computer systems: the complete "mirroring" of the information such as fixed or variable data. In addition to the configuration data and the dynamic data, (call status, call detail records, traffic collection, etc.) the complete set of programs and software modules must 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 of communications 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

D.

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 working order in a third system and which will be stored automatically in the Buyer organization's computer back-up system. E. Security: Dedicated technical areas are locked, controlling access to the system and to the cabling system cabinets. The system must offer standard protection for password control features as well as system access by password / shadow password file / aging passwords. In addition to this basic rule, the system must incorporate these other security aspects. 1. Controlled access to management platform: One of the most important security considerations is protecting access to system management. The system must control the identity of the management terminals and the user accessing that terminal. During a connection, (local or remote) the system must check the consistency between the management platform name, management platform password, and user name before authorizing the connection.
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2.

Secure remote access via a public switch exchange: The system must offer several secure management and maintenance access through the public switch exchange. The remote access proposed, either via PSTN or ISDN, must provide a high security level for remote management/maintenance terminals at a predetermined location, as well as the normal user name and password control. Remote access through Internet or Intranet: A firewall is the only mechanism that can adequately protect the computing and communication resources via Internet or Intranet access. But to protect and control remote access through IP networks, the proposed system must support a filtering engine to control IP devices and services that have access to the system; a light and cost effective embedded protection mechanisms such as a trusted host and TCP wrapper.

3.

F.

IP Communications 1. IP station support: The system must manage, control and support a range of IP telephone stations for both voice and telephony applications as well as IP application stations for voice, telephony, and Web services support. IP telephone and IP application telephone stations will be referred herein as IP stations, and when necessary, the type of IP station will be clearly stated. Quality of Service: The proposed system supports native IP communications in direct or "peer-to-peer" with only the telephone signaling transiting back toward the controlling communications server. The speech will be switched over the IP network and exchanged directly from client to client. The voice and signaling frames should be marked [tagged] in order to be recognized and should be classified by the network. The standards of marking supported will be: Level 2: IEEE 802.1p /Q Level 3: TOS / DiffServ

2.

In the case of a PC connected to an IP station, (IP telephone or IP application telephone), the frames transmitted by the PC, tagged or untagged, must be treated 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 the terminal) by the Clients compatible DHCP within their server. Automatic VLAN Assignment: Although voice and data traffic flow over different VLANs, they will be managed simultaneously because of their distribution over one network. When a user station is moved, its possible to perform the IP activation for a VLAN voice connection that is different from the one initially programmed in the terminal. In this case, the telecommunications system should have the capability to support a procedure, based on standard mechanisms, to assign the corresponding VLAN number to the IP station clients during IP station initialization. IP Communication Zones: An IP communication zone (IP domain) is defined by IP stations and / or media gateways within a geographical area.
16720-6 Telephone System

4.

5.
ITCC in Riyadh Residential Complex J10-13300

Because bandwidth is shared by the data and voice communications, its necessary to limit the bandwidth assigned. The goal is to take into account the underlying IP infrastructure to avoid congestion in the WAN links and maintain good quality voice communications. The system must be able to control and limit the number of simultaneous communications between the different communication zones. In addition, the compression algorithms used for the communications inside the same IP communication zone and between different IP communication zones can be different and must be activated on a call-by-call basis. For example, inside the same IP communication zone (same LAN), communications will not be compressed but digitized in G.711 in order to have toll quality voice; whereas inter-zone communications will use the compression algorithm defined by the administrator: G723.1 or G.729A. 6. System Availability: In the distributed IP architecture with a soft switch controlling the IP stations and the media gateways, their operation must be maintained by an emergency arrangement. In case the IP network goes down, the link with the soft switch must recover automatically via the analog public network or ISDN. In this case, the media gateway will behave like an autonomous PBX, and if some IP stations are located on the same site as the media gateway, they will also recover their signaling to the communication server through the MG. H323 Compatibility: The proposed telecommunications system must support 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)

7.

Gatekeeper Server: The proposed system should integrate an H.323 gatekeeper 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 DHCP server 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 registering itself with these entities. H.323 Gateway: The proposed system should include a gateway that allows the Buyers H.323 devices to interoperate with the traditional telephony devices (digital stations, IP, analog, private or public lines) and the SIP terminal. The main H.323 gateway features required are:
ITCC in Riyadh Residential Complex J10-13300

Support of the H.225 and H.245 protocols "Fast-connect" setting H.245 tunneling Registration, admission, status (RAS)
16720-7 Telephone System

 8.

Direct mode and routing mode Connection to an external gatekeeper H.323 attachment D for T38 fax Direct RTP

SIP compatibility: The proposed system should permit the integration of SIP terminals with other terminals and private or public external lines used by the company. The SIP software should conform to the normalized architecture and be integrated in real-time communications management to benefit 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 supported standards 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.txt o 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 to register itself on the system via the SIP proxy module AND to see itself. Its also assigned a directory number in the communication server so that traditional telephone services are also available. The following services are required 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
16720-8 Telephone System

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 10.

Call records for external calls

Authentication: An HTTP Digest (MD5) authentication procedure must be defined 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. External SIP communications: The SIP terminals in the organization, as well as the traditional stations should be able to communicate with facilities behind a SIP connection or external SIP proxy (service provider class or enterprise). All SIP calls must pass through the system SIP proxy, which controls the maximum number of calls. It must also be possible to arrange interception and prohibition of calls from unknown SIP terminals to all Client stations.

11.

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 for integrating telephony services into web applications. The solution must be able to handle a high capacity of clients using XML services. The access to the XML telephony application must be protected by user login name and password . The following XML telephony features are required to be available to users equipped with one of the following phone types: analog, digital, wireless (DECT/PWT), and IP telephone stations. Login / logout Password Telephony 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 web services from corporate or external Web application servers must also be XML compatible. H. Connectivity interfaces: The telecommunication system will support the following connectivity: Public Switched Telephone Networks (PSTN):
ITCC in Riyadh Residential Complex J10-13300 16720-9 Telephone System

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 2.2 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

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 new stations in future.

2.3

SYSTEM FEATURES A. Unified Communications: The system will offer a complete suite of telecommunications applications that enable users to control and manage their calls, voice messages, email, fax, directories, and collaboration tools. Unified access to these applications will be possible via an HTML interface, or through
16720-10 Telephone System

ITCC in Riyadh Residential Complex J10-13300

the use of a telephone terminal. B. The system shall include the following features: 1. 2. 3. 4. 5. 6. 7. 8. 9. 10. 11. 12. 13. 14. 15. 16. 17. 18. 19. 20. 21. 22. 23. 24. 25. 26. 27. 28. 29. 30. 31. 32. 33. 34. 35. 36. 37. 38. 39. 40. 41. Last number redialing Company directory Music on hold Hands free telephone sets Paging interface Manual Message waiting Display the dialing number and intercom calling station number for some sets Ring again Follow me Facilities of tie lines with other PABX Facility of using extension lines as data lines for computer Classification of extensions Extension to operator calls Outgoing calls Booking outgoing calls Operator to extension calls Inquiry calls Transfer calls Conference calls up to 8 parties per call Priority Common night service Universal night service Individual night service Direct in lines Automatic call back Individual call diversion Individual call diversion when no answer Flexible numbering Line lockout Choice of individual trunk line Group hunting Control of long-distance dialing Direct speech connection Call pick up Grouping of trunk lines Call waiting indication to extension Common abbreviated dialing Hot line Trunk to trunk transfer Calling names numbers display on digital instruments Incoming external calls with: Camp-on busy Break-in and forced release Operator recall 42. Additional Facilities: Push-button dialing Night time classification of extension
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Telephone System

2.4

ATTENDANT (OPERATOR) POSITIONS: A. Propose a range of attendant stations for Engineers approval, with a precise description of the features provided by each of them. Attendant terminals are required to have a color screen, as well as attendant station software that can be used on PCs that are not dedicated to the attendant operation. The answering device will be designed specifically for answering calls; the presentation of calls; chaining of calls, and re-dialing will be entirely controlled by the communications system. Nevertheless, it will be possible to hold specific individual calls on keys reserved for this purpose. The capacity of the different waiting queues should be unlimited. Signaling or indications displayed on the terminal should be sufficiently explicit and unambiguous to facilitate the handling of 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 of the terminal, etc.) Propose a device that allows the operators of the attendant positions to continuously supervise (view) the state of some stations in the installation (free, busy internal, busy external, unavailable). There must be at least 40 supervised terminals. The attendant will be able to modify the mode of answering calls at their answering console. In automatic answer mode, calls coming into the attendant console will be presented and connected without requiring manual intervention by the operator 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

B.

C.

D. .

2.5

TELEPHONE TERMINALS (HANDSETS) A. General: Submit for Engineers approval a range of digital telephone equipment as well as IP stations, each capable of meeting one or more of a variety of needs, ranging from basic equipment for use in public areas to that of business or management class. Type and number of telephone terminals shall be as shown on the drawings or as approved by Engineer. Analog Terminals: The analog terminals shall be able to respond to either pulse or DTMF dialing signals, and automatically recognize the type of signal being received. Analog Terminals with Caller ID: Any standard analog Call ID telephone from well
16720-12 Telephone System

B.

C.

ITCC in Riyadh Residential Complex J10-13300

known manufacturer suitable for interfacing with the system. The terminal shall show the incoming internal and external caller ID in case CLI information is available on analog/digital/ISDN trunks. The analog terminals shall be able to respond to either pulse or DTMF dialing signals, and automatically recognize the type of signal being received. D. Digital Telephone Stations: Propose a range of digital terminals for Engineers approval and provide precise descriptions of the features available with each of them. Only electronic digital terminals will be accepted. For each type of terminal proposed, 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, in the following configurations: E. Common or different extension number Analogue interface dial independent of the host terminal Analogue interface dial for the host terminal

IP Telephone Stations: Propose a range of IP telephone stations for Engineers approval. These IP telephone stations shall be identical to the digital stations in aspect and function. For every type of station proposed, specify if the stations support the following features: 1 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 Type 1: Executive Telephone (Supervisory or user positions handling heavy traffic)

Executive IP Phone supports a rich suite of endpoint applications, including XMLbased applications on all models. Selected models also support MIDlet-enabled applications, Unified Communications Widgets, applications developed within the Unified Application Environment and personal video communications. High-definition (HD) voice, vibrant color displays, Gigabit Ethernet connectivity, and more than basic support for endpoint applications. Executive IP Phone supports multiple-call per-line appearance, users can take advantage of advanced call navigation capabilities with support of multiple-call
ITCC in Riyadh Residential Complex J10-13300 16720-13 Telephone System

sessions on a per-line basis. For example, on a 2-line endpoint, you can be on an 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 and forth between these two call sessions, as required. Executive IP Phone supports a rich suite of endpoint applications, including XMLbased applications on all models. Selected models also support MIDlet-enabled applications, unified communications widgets applications developed within the unified application environment and personal video communications. Features: 2 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 application detail and interaction High-definition voice (HD voice) support of headset, handset, and full-duplex speakerphone for superior audio performance Integrated IEEE 10/100 switch ports support co-location of a PC at the 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

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 large volume 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 greater convenience 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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ITCC in Riyadh Residential Complex J10-13300

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

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 to moderate voice communications usage. Features: - Half-duplex speakerphone and internal microphone for clear communications - 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 application stations full-featured with integrated IP connectivity and telephony, capable of supporting any web-based business application that is compatible with XML. At least 2 IP application stations shall be provided, unless indicated otherwise For each type of terminal proposed, provide precise descriptions of the features available with each of them as well as specify if the terminals support the following 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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ITCC in Riyadh Residential Complex J10-13300

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 a set of tools customizing their communications to the specific day-to-day demands of work and to fit specific enterprise, group and individual needs. 2.6 POWER SUPPLY A. Power Supply: Device for controlling the designated minimum and maximum values of the continuous power delivered. When one of these thresholds is reached, the energy feed to the telecommunications system must be cut automatically. Control devices will provide both automatic and manual control of the float and balancing, with automatic return to float mode when the battery has been charged to the required level. The rectifier and power supply will be sized to feed the power requirements of the telecommunications system at the designated cabled capacity, and should be able to recharge the batteries within 10 hours Battery: Batteries in the power supply will be the maintenance-free type. Batteries will be installed to power the telecommunications system automatically in case of a failure of the electric power or the rectifier. This uninterruptible power supply will maintain the telecommunications system at full traffic for a minimum period of four hours. Distributed Power Architecture: For an IP network with a distributed architecture where the media gateways power is connected directly on the sector, an UPS
16720-16 Telephone System

B.

C.

ITCC in Riyadh Residential Complex J10-13300

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 the latest technologies, such as JAVA/JEE. Provided at least one management server, unless indicated otherwise. The server shall support a minimum of five (5) clients having different access rights to the applications. Access to the server must also be possible by a light Netscape or Microsoft Explorer client. The management 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 modification to the telephone data will automatically generate a change in the telephone data, and vice versa, any modification of the user station parameters will generate an update in the directory. For example, all directory data that is modified will automatically update the telephone system and reciprocally, all subscriber parameter changes will update the directory. A user shall also have a complete station information in a window: B. Management parameters Information on communications costs Information on telephone traffic

Connectivity and Modem Access Security Management: The management server must be able to be situated anywhere on the premises, and accessible to the communications system(s) via an IP network, the PSTN, or ISDN. The telecommunications system must be able to control remote access coming in via the PSTN or ISDN. In an ISDN (CLIP) environment, the system must be capable of authenticating the identification of the requestor transported by the network. If the number does not correspond to an authorized access, the request is denied and an alarm is generated, indicating the calling number information for the unauthorized request. In the PSTN environment, when the network does not transport information, the remote (CLIP) must present a user name and password for the connection. If the identification is correct, the system releases the call and calls a predefined number in memory that corresponds to the user name. Management Access Security: The access to the services offered by the management server should be protected in the following manner: Authentication of the management server Authentication of the client user

C.

Authentication of the server and customer should be based on a user name and password. During the connection phase, the voice communications system should test the coherency of three items:
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Telephone System

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, it should use IPsec mechanisms between the server and the management clients. The management server authenticates the user password in order to permit domain management. D. Domain Management: A primary administrator who has access to the suite of applications and information residing in the management server (including masked numbers and personal codes) can define domain management. This primary administrator will have the right to create user profiles defining the following: E. 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

System Management: The telecommunications system configuration and management application must permit management of all the parameters via a graphical user interface. The configuration application should have the following features: 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 data regarding system traffic, use of telephone features, and overall system operation at all times. Collection of traffic data is required for measuring system use and to detect possible system sizing problems. The application results must be presented in a graphical format. Bidder will describe and propose the various traffic 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 that permits the reloading and subsequent use of the information. G. Communications Cost Management: Call record storage on the system hard disk shall be for a minimum of 100,000 call records. Management and sorting of these tickets will be performed using the management and traffic analysis application residing on an external server. Call records will be automatically downloaded by the application at night; for security reasons, a sliding file of the most recent seven (7) days will be maintained continuously on the system hard disk. General Characteristics: On ISDN connections, the proposed system will not use direct or per-call costing means from the carrier. On calls transferred by a station or attendant, outgoing call costs should be assigned in a manner that accurately reflects the amount of time each party was in control of the call. For reasons of organizational confidentiality, some stations will never be tracked for any reason; the costs generated by these stations will, nevertheless, be reflected on the cost counters. If it is not possible to receive cost reference information from the carrier, the call accounting management system must be able to accurately calculate communications expenses by an alternate means. The application must be able to handle up to five carriers (service providers) simultaneously; a simulation utility integrated into the application will permit the system to use the stored information to identify the lowest cost service provider for external traffic. Indicate the various types of information storage and will specify the following elements: 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.

H.

To facilitate the overall management, automatic updating of the application is required 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 permit multiple organizations with up to eight levels each. Access to each of the levels in an entity will be granted to multiple users as needed (department director, general manager, etc.) by the system administrator. These users will be assigned an access code that provides access to all the system features (such as creating reports, program a prerecorded observation, etc.) for their organizational grouping. The ability to manage as many as 10 users, in addition to the primary administrator, shall be part of the feature. The hierarchy information must be automatically synchronized when the telecommunications system undergoes a relevant change (stations, attendant(s), trunking, cost centers, etc.) to reflect the
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ITCC in Riyadh Residential Complex J10-13300

organizational structure. The hierarchy information must be printable. Call detail records for cost reports: For every outgoing call, the system shall provide 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-defined reports, in addition to a management report utility to customize the existing reports or to create completely new ones. It will be possible to create graphs from these reports, in pie and histogram format. The following points will be determinants 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 email advice should be sent to one or more administrators. J. Management of the System Alarms: The management server should offer an application that centralizes the alarms and relevant systems communication events, as well as any generated by the management server itself. These events must be filtered and displayed in real time, according to the administrator's needs.
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These alarms should be categorized according to the six-level ISO severity definitions, and identified by different colors to facilitate direct attention to specific categories. Every alarm should have a detailed explanation at the second level; in addition to likely causes, the application should indicate the appropriate troubleshooting actions to resolve the problem. In the case of a major alarm, an email must be automatically sent to one or more system administrators. The list of alarms and events, as well as their detailed explanations must be printable and archive-able. A report generator will provide statistical reports. K. Topology (For multi-site application only): The management server should provide an application that offers a topological view of the telecommunications system that constitutes the organization's network, as well as the links that exist between sites. The presentation should be the simplest possible, and consider the need for an automatic discovery utility that will automatically display the subnetworks and nodes that exist. This application must be completely customizable in terms of the screens and the icons that represent the objects. The different objects that represent the organization's network must be synchronized with the alarms application in real time, and items that are affected by alarm-generating conditions must indicate the event, using the same severity and color format. When a problem appears, the administrator will access the faulty element by a mouse-click, and will move through the system architecture, following the network hierarchy (node, cabinet, media-gateway, board, etc.) until the problem element is found. The administrator will be able to automatically launch system configuration 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 it by 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 messaging application. An integrated system is preferred because it provides the best possible inter-operation. Describe all the services available to the users and the voice messaging system administrator, the maximum number of voice mailboxes, the maximum recording capacity, and number simultaneous links that can be handled. In addition to the required services described below: 1. Answering or answering with date stamp: The system shall provide voice mailbox holders the choice of two functions: answering the messages or answering them with a date stamp. Customizable Announcements: When a call is forwarded to the voice messaging system, the box holder will be able to choose between two personalized announcements. If the personal announcement has not been
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2.

ITCC in Riyadh Residential Complex J10-13300

recorded, the automatically. 3.

standard

system

announcement

will be

substituted

Call forward to the voice messaging system from a box holder: When a box holder agrees to have his or her call forwarded , or after a set amount of time that there is no response, the internal or external caller is automatically directed to the voice messaging system. After the caller hears the announcement, the caller can record a message or decide to be forwarded to an attendant or a specific station. Overflow of Internal Calls: After a set amount of time, the system will offer voicemail to internal callers for those who have a voice mailbox. Recording of Calls: The holder of a voice mailbox will be able to take advantage of this service to record internal or external calls. Recorded calls will receive the same service as messages that have been left by callers. The minimum recording length will be 10 minutes; This service shall be automatically provided to digital station users. Message waiting Indication: The box holder is advised, by a lit lamp on the station (either digital or analog equipped with a message lamp) that there are messages in the mailbox. Users who have terminals without a message lamp should be advised by a voice prompt that there are messages waiting, when user picks up the handset to make a call. Message Review or Access: The box holder can review or access waiting messages from any internal station or an external station via the general telephone network. When using a digital terminal, the display will offer dynamic context-sensitive soft keys that operate the message system features on an interactive basis: fast forward, stop, rewind, storage, continue, previous message, etc. Access security: Security and privacy for the recording of personal announcements and in the review of messages will be assured by a personal code. Spoken Newspaper: A spoken word "newspaper" or "information" service with a minimum length of 30 minutes should be available. Broadcast Lists: The box holder will have access to general and personal distribution or broadcast lists. Forwarding of Voice Mail Messages: The box holder will be able to send a copy of previously received messages to other boxes (with or without requesting acknowledgement of receipt). Message Storage: The messaging system will store messages automatically, for a programmable period of time, after which the message will be automatically erased; only a specific action to erase a message will override the storage of the message, and erase it immediately. Call by Name: To provide universal access, it must be possible to select a voice 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 as available to an outside caller, in which case, the caller will be given the
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4. 5.

6.

7.

8.

9. 10. 11.

12.

13.

ITCC in Riyadh Residential Complex J10-13300

choice to deposit a message or having the system call the desired party directly. 14. Multiple Languages: To ensure consistency with the system voice prompts, the proposed system should be multi-lingual, offering four different languages (as selected by Client). The service voice prompts played to a user will follow the same language the user has selected in the telecommunications system. In no case will a mix or combination of languages on the same voice prompt be accepted. Automated Attendant: The messaging system should offer an integrated and interactive automated attendant service. The application of messaging should 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.

15.

2.9

RACEWAYS , WIRES AND CABLES: A. B. For raceways/trunking refer to Section 16050 - BASIC MATERIALS AND METHOD. Other Wires and Cables: Refer to Section 16715 VOICE AND DATA COMMUNICATION CABLING.

PART 3 - EXECUTION 3.1 INSTALLATION: A. B. C. D. 3.2 The entire system shall be installed by specialist subcontractor approved by the Engineer. Installation shall be in accordance with the approved drawings and manufacturer's written instructions. The incoming external cables will be jointed after entering the building with dry plastic type cable prior to connection to the MDF (Main Distribution Frame). For excavation and backfilling refer to Section 02200 - EARTHWORK.

ELECTRICAL SAFETY: A. Separation of telecommunication circuits from the building electrical system and electrical equipment shall conform to the latest publications of Articles 800-3 (a) and 820-13 of National Electric Code (NEC). An earth or ground shall be provided and extended to the termination box and connected to each station protector. The earth or ground shall be installed and bonded in accordance with Article 250, 800 and 820 of National Electric Code (NEC) and conforming to Section 16452 - GROUNDING..

B.

3.3

TESTING: A. After installation of entire system and prior to acceptance of work, manufacturer's standard 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 Saudi Telephone Company for telephone connections and provide them required information and help to complete all paperwork. Charges for telephone connection (by Saudi Telephone Company) shall be borne by the Owner. END OF SECTION

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