Single System PBX Network: On-net Multilocation Support

The simplest network design consists of a single PBX system. There are several single system design configurations based on the location of the system’s station users. These configurations are used to support the following types of communications requirements:

  1. One or more station users at the same location who are physically remote from all customer premises locations housing PBX common equipment

  2. One or more station users residing on the customer premises but exceeding the maximum loop length for their desktop telephone equipment

  3. Station users physically remote from the customer premises location housing the main PBX common equipment room but supported by local common equipment

The first PBX configuration option supports remote teleworkers. The second and third configurations are based on a distributed common equipment architecture. Some PBX systems have a standard distributed port interface cabinet/carrier design, and others use hardware options designed for remote premises communications requirements.

The first single system PBX networking category supports station users who are remote from all common equipment but require desktop communications support as if they were located at their organization’s premises. These type of station users are now referred to as teleworkers. Teleworkers fall into two categories:

  1. Fixed teleworker

  2. Mobile teleworker

A fixed teleworker has a permanent desktop location relative to the remotely located PBX system. Fixed teleworker solutions are sometimes referred to as small office/home office (SOHO) configurations. Mobile teleworkers are constantly on the move, but need to be linked to their PBX system whenever and wherever they are. Another popular term for a mobile teleworker is road warrior. Road warriors are the newest breed of PBX station user and are growing in number at an alarming rate due to the proliferation of mobile computing and communications devices.

There are several available PBX options that support fixed teleworkers who require the same level of communications service and support as station users at the customer premises location. The oldest, and most basic, PBX option supporting off-premises station users is the OPX feature, a solution available for more than 20 years. The OPX feature requires a special local exchange carrier trunk circuit, known as an OPX circuit, to provide PBX system communications and signaling support to a remote analog telephone. The loop length of an OPX circuit connection to the remote station is usually limited to several miles without repeaters. Provisioning of OPX trunk services is also constrained by LATA boundaries. OPX station users have full access to all PBX features and functions but are operationally limited by their analog telephone instruments. No analog telephone supports out-of-band signaling to support multiple line appearances and proprietary PBX display field information. Off-premises and intercom calls to the remote station user are routed over the PBX’s OPX trunk. The remote station user can initiate intercom calls to other PBX station users and uses PBX trunk facilities for placing calls outside the PBX system.

SOHO applications requiring a higher performance desktop terminal, similar to the one available at the customer premises location, had limited options until the early 1990s. During the past decade several fixed teleworker solutions have been implemented:

  1. Remote ISDN BRI telephones

  2. Proprietary and third-party local loop distance extender options

  3. Remote IP telephones/softphones

The first SOHO option based on ISDN BRI services was implemented by Intecom in the early 1990s. The PBX system interfaces to the PSTN via a digital trunk circuit by implementing ISDN PRI services, and the remote teleworker uses an ISDN BRI line to support a desktop ISDN BRI telephone that offers performance capabilities comparable to those of proprietary digital telephone instruments. PBX control signaling is carried over the ISDN trunk circuit’s D-channel to the remote location. This solution provides remote teleworker access to all PBX system features and functions and can support remote data terminal equipment over the second ISDN BRI B-channel.

A popular SOHO solution has been the use of local loop distance extender equipment to support proprietary digital PBX telephones over the PSTN trunk carrier facilities used to transport voice communications and control signaling between the customer premises and remote location. The first distance extender options were based on hardware equipment at the customer premises location—the proprietary port circuit cards or gateway modules, used to convert the proprietary out-of-band digital desktop control signaling into CAS format for transport over analog trunk circuits to connect PBX and remote teleworker locations. A desktop gateway module at the remote teleworker location supports a proprietary desktop digital telephone. Leading suppliers of local loop extender equipment include MCK Communications and Teltone. The option was developed when residential digital line services, such as ISDN BRI, were not commonly available. The growing availability of digital subscriber line (DSL) services has eliminated the need to convert digital signaling to CAS format. The wideband nature of current digital services available to SOHO locations allows several remote digital telephones to be supported over a single DSL line. Rack-mount carrier gateways at the remote location can support 12, 24, or more digital telephones with one or more digital T1-carrier facilities.

One of the very few early benefits of IP telephony was support of remote IP telephones using dial-up analog or digital lines at a remote location. Almost all IP-PBX systems can remotely support desktop IP telephone instruments or IP softphones via LAN/WAN infrastructures. The remote location configuration may require an analog gateway module or a SOHO router to access the corporate WAN.

Road warrior options include IP softphone applications running on a notebook/laptop computer, or any DTMF analog telephone or cellular handset, linked to the main PBX system through a proprietary communications/signaling port interface card, gateway module, or teleworker server. The mobile IP softphone option can be implemented by local LAN access to the corporate WAN or a wireless dial-up option wherever the station user happens to be. Several PBX suppliers are currently marketing mobile telephone options behind their communications systems, including Avaya, Nortel Networks, and Siemens. The Avaya Definity and Nortel Networks Meridian 1 options are based on the MCK Communications Mobile EXTender gateway. The EXTender gateway is configured behind the PBX system with the use of standard digital port interface links. The Siemens solution is based on a HiPath Teleworker server linked to the HiPath PBX system. For each road warrior option, the remote desktop telephone or cellular handset appears to the PBX system as an extension of the station user’s customer premises desktop voice terminal. Any DTMF telephone can be logged into the PBX to receive and place calls through the centrally located communications system. Road warriors can use intercom; four-digit dialing plans; activate basic call processing features, such as hold, transfer, and conference; and use the private PBX network for long distance calling. Unanswered calls are routed to a call coverage station or VMS mailbox rather than the remote telephone’s mailbox.


Private Networking

If the ARS feature determines that a placed call is to be routed on-net, the call is handled over private network facilities. There are many private network configurations, ranging from support of a single station user working at home, to linking of dozens or hundreds of PBX systems and locations scattered across the globe. Private networking options can be classified according to the following criteria:

  1. Number of PBX systems in the network design

  2. Number of station users at each network location

  3. Locations of station users relative to a local PBX system

  4. Available transmission carrier facilities

  5. Traffic capacity requirements between network locations

  6. Feature transparency requirements across the network

  7. Survivability requirements for each network location

  8. Operations, management, maintenance, and service procedures


PBX Networking: CBCSS

CBCSS is a PBX network feature that allows a single ISDN PRI services trunk group to carry calls to multiple network carrier services or facilities or carry calls using different interexchange carriers. CBCSS uses the same routing tables as those used by ARS/AAR. Without CBCSS, each trunk group must be dedicated to a specific carrier service or facility. Implementing the CBCSS feature allows a variety of services to use a single trunk group. The services are specified on a call-by-call basis. This optimizes trunking efficiency because traffic is distributed fully over the total number of available trunks, regardless of peak time period service requirements. Examples of services typically requiring dedicated trunk groups without using CBCSS are in-bound and out-bound WATS; direct long distance dialing; VPN; digital data services including ATM, frame relay, and IP; digital video services; presubscribed common carrier operator international 800 calls; and other user-defined services.

CBCSS allows services to share the same trunks to reduce total trunk requirements. The probability of denied feature and service access due to blocked trunk access is also reduced. Network engineering is simplified because trunk engineering analysis can use total traffic data instead of analysis by a per-service basis. By dynamically changing the mix of trunk circuits accessing different services and facilities, the system can function more efficiently.

CBCSS customer programming tools allow administrators to dynamically assign and reassign individual trunk members in a trunk group access to different services and facilities based on pre-set schedules (time of day, day of week), real-time traffic loads, or on demand.

The NFAS feature allows an ISDN PRI DS1 interface D-channel (signaling channel) to transmit signaling information for B-channels on ISDN PRI DS1 facilities other than the one containing the D-channel. A single D-channel can carry signaling information for numerous B-channels on different DS1 carrier facilities, thus providing a more economical interface between the PBX system and the ISDN network. This means that a customer can configure a single D-channel to support more than the standard 23 B-channels available on a facility-associated signaling ISDN PRI DS1 carrier circuit. A single D-channel can therefore provide signaling for 50, 100, or more B-channels based on the software programming limitations of a PBX system supporting the NFAS feature.

PBX systems implementing NFAS also can support D-channel backup. If a D-channel signaling link fails, a backup D-channel transports the signaling. The feature requires that one D-channel be administered as the primary D-channel and a second be administered as the secondary D-channel. When a transition from one D-channel to the other occurs, all stable calls (calls already answered) are preserved. Some messages may be lost, resulting in a loss of call-related information, but the calls themselves will be maintained.


PBX Networking: ANI/SID

ANI is delivery of the originating calling party’s 10-digit billing telephone number to an ISDN subscriber’s premises communications system and/or desktop terminal equipment. When a telephone call is made from an equal access CO, the ANI is passed from the local exchange carrier network to the interexchange carrier network for transport across Common Channel Signaling System 7 (CCSS7) packet switched facilities to the destination local CO exchange. The ANI is included in the call setup message using Q.931 message format over the D-channel. ANI is delivered with ISDN BRI or PRI trunk services.

ANI is an interexchange carrier service often confused with local exchange carrier CLID service. CLID is a local access transport area (LATA) service feature that is one of the commonly available customized local access signaling system (CLASS) features. ANI is a service feature offering that was originally developed in support of long distance telephone calls placed to inbound call center operations, with a requirement for ISDN trunking to the customer premises. PBX/ACD systems receiving the ANI can use the information for call screening, analysis, decision making, and routing procedures, including a database lookup procedure to match the ANI with a customer file. ANI also can be used to identify the geographic location of the calling party, because the area code is included in the digit string. ANI is no longer used exclusively for call center applications but is considered an important element for enhanced call screening functions at the system or desktop level. Another useful ANI benefit is preidentifying the calling party for network security purposes.

An ISDN feature similar to ANI is SID. SID is usually more useful than ANI because the feature delivers the originating caller’s telephone number behind a PBX system. ANI is the trunk billing number of the PBX trunk circuits and does not identify individual callers provisioned behind the switching system. SID can distinguish station users to a greater degree than ANI.

ANI and SID are useful features for a variety of customer applications that do not include formal in-bound call center systems. Collecting ANI/SID data allows PBX customers to better track and analyze incoming long distance calls by geographic area (regional and local). Collecting and storing ANI/SID in-bound call data in a call log database can support improved out-bound customer service and telemarketing operations. PBX customers can use a variety of network carrier services to route incoming calls to different PBX systems by using ANI/SID data to load-balance calls across locations for increased call handling efficiency and performance.


Public Networking: ISDN Features

ISDN evolved from the original analog-based telephony PSTN that provided end-to-end connectivity to support a wide variety of services to which users had access through a limited set of standard multipurpose customer interfaces. The major attributes of ISDN include:

  • End-to-end digital connectivity

  • Access and service integration

  • User control of service features

  • Standardized user-to-network interfaces

The CCITT (later the ITU) originally defined two major, globally standard interfaces: BRI and PRI. The BRI provides a station-direct interface to ISDN network facilities or ISDN-compatible customer premises equipment. The PRI provides an interface for switch connections, such as PBX-to-PBX. Both interfaces have similar format structures: one D-channel and multiple B-channels. The B-channel, or bearer channel, is a 64-Kbps transmission path for basic and primary rates and carries voice, data, and image or video communications traffic to and from the network or switch. The D-channel, or data channel, uses 64 Kbps for primary line and 16 Kbps for basic lines and carries packeted signaling and control information across the interface. The D-channel signaling is out-of-band because it supports calls on the separate B-channels. In North America, the PRI has 23 B-channels and 1 D-channel (23 B+D); outside of North America PRI has 30 B-channels (30B+D). ISDN services are carried over T1/E1 transmission circuits using the DS1 format structure: (24/32) 64-Kbps channels. Globally, the BRI has 2 B-channels and 1 D-channel (2B+D).

The ITU standard established for D-channel signaling is the Q.931 message-oriented signaling protocol. This protocol allows the use of the same access lines for many different services and the introduction of new services without requiring users to replace existing ISDN-compatible communications equipment. ISDN is defined internationally as the physical interfaces between communications equipment and networks and the signaling protocols exchanged by the elements.

Several PBX network features that work with ISDN PRI services, such as ANI and Station Identification (SID), CBCSS, non-facility associated signaling (NFAS), and D-channel backup.


Public Networking: Automatic Route Selection (ARS)

All station user calls, direct distance dialed and private network, are routed to trunk groups that have access to exchange carrier facilities terminating in a central office. The many types of port circuit interfaces and trunk facilities. The software feature controlling access to the trunk groups is ARS. PBX ARS features and functions support control and routing of calls over public network carrier facilities and across private networks. ARS is used to select among various types of trunks:

  1. Local CO, analog or digital

  2. Foreign exchange (FX)

  3. Wide area telecommunications service (WATS)

  4. Tie line (private line); analog and/or digital

  5. ISDN PRI service circuits

When the ARS software program routes calls to public network carrier facilities, it is referred to as off-net routing; when calls are routed to private network carrier facilities, it is referred to as on-net routing. Off-net calls are based on the public network dialing format and terminate in the public network. On-net calls are based on a private network dialing format and usually terminate within the private network, although some calls may be routed off the private network and onto public network facilities, if the called station is not configured as part of the network. The private networking feature used to describe the latter routing option is known as tail end hop off (TEHO). On-net routing programs support tie-trunk protocols, proprietary private network protocols, and industry-standard private network protocols.

The ARS feature is activated when a station user dials the access code for an off-premises call requiring a trunk circuit, followed by the telephone number to be called. Public network calls in North America are usually dialed beginning with the digit 9, which alerts the common control complex that an outside line (trunk) is required to place the call. Calls placed over traditional private tandem networks are usually dialed with a customer-selected digit, such as 8, to distinguish the call from public network calls. Customers with traditional private tandem networks create a unique numbering plan for on-net calls. An intelligent private network call is placed by dialing a station directory number within the network that matches the local directory number of the dialed station. Intelligent private networks do not require a separate private network numbering plan because the entire network of PBXs operates as a single homogeneous network for most system features and functions, including the uniform dialing plan. Small network configurations may be supported using a four- or five-digit numbering plan; larger networks may require more dialed digits for on-net calls because a limited digit dialing plan cannot support the number of station users.

ARS routing table rules determine whether the call is routed off- or on-net. For off-net calls the ARS program determines which trunk group the call is routed to for network access. PBX system administrators rank trunk groups from lowest to highest cost to ensure that calls are sent over the least costly route available to the caller, based on customer’s class of service or restriction level. Public network route selection is based on data in the dial plan databases. The ARS software analyzes and compares the dialed digits with the digit string patterns in the ARS dial plan. If there is no database match, the call is blocked. Blocked calls may be the result of call restriction levels for individual callers. It is common today for system administrators to block many outgoing 900 calls, or direct-dialed international calls, except for select station users. Restriction features may limit an individual station user’s dial capabilities to internal calls or calls to very select off-net locations, such as local exchange calls.

If there is a database match, the system may request an account code before continuing, or the ARS feature can immediately define the call route and determine which digits should be sent over the network while the call processing software seizes an available trunk according to the station user’s network class of service level. If all trunk circuits are busy for the lowest-cost trunk group, the call may be routed to the next highest-level trunk group based on call costs, or the call may be queued until a trunk circuit is available in the originally selected trunk group.

The trunk circuit seized by the PBX system is determined by analysis of the ARS routing table and associated trunk tariff database. The ARS routing table is made of routing patterns that map to call routes for the dialed CO location code. CO codes are defined by specific country and city locations. For each dialed CO code, there may be one or more routing patterns, and each routing pattern may have one or more call routes. More than one pattern of dialed digits can translate to the same routing pattern. The facility restrictions level (FRL) feature determines access to a select call route within a routing pattern. For any particular call route, multiple trunk groups may be used to handle the call, if the customer subscribes to more than one exchange carrier service. The lowest-cost route is usually programmed as the preferred route if the trunk circuits are available.

The most common ARS feature capabilities are:

  1. Area code/office code restriction (toll restriction)

  2. Alternate route selection (route advance)

  3. Time-of-day routing

  4. Day-of-week routing

  5. Trunk queuing

  6. Digit analysis and manipulation

  7. Call screening


Distinct IP Telephony Features/Functions

Integrated port interfaces. Compared with a legacy digital telephone, an IP telephone can be designed and equipped to provide several unique feature/function capabilities. An IP telephone design attribute not available with traditional digital telephones is the integration of a multiport Ethernet hub/switch to allow multidevice sharing of a single connector port to the Ethernet switched network. Most current IP telephone models are equipped with two Ethernet port connectors: one connector for the Ethernet network and one for a desktop PC client. Mitel Networks has indicated that its next-generation models will have another external connector port to support two Ethernet devices external to the telephone. An integrated Ethernet port interface reduces telecommunications outlets, inside wiring, and Ethernet switch port requirements. Cisco Systems was the first supplier to incorporate an integrated Ethernet switch into its IP telephones in its 7900 series. Mitel Networks followed Cisco’s approach by including integrated Ethernet switch ports in its second generation of IP telephones. Avaya, still marketing its first generation of IP telephones, offers its IP telephones with an integrated Ethernet hub.

The difference between an IP telephone with an integrated switch or hub may not be important to most customers, but providing a high level of voice-grade communications to the desktop is of primary importance. Voice communications QoS at the desktop can be supported using a variety of methods, such as Ethernet LAN 802.1 p/Q, or CoS programming (by switch or hub port). For example, each Cisco 7900 IP telephone internal Ethernet port can be programmed for different classes of service; the default service level of the voice port is a 5 and the data port is a 0. The system administrator can override the default service levels, if required, by an individual desktop station user. IP telephones with an internal Ethernet hub must include customized software to prioritize voice communications.

Besides Ethernet port connectors, current IP telephones may also support peripheral data devices through a USB port or infrared interface to a PDA. A USB port theoretically can be used for a variety of devices, such as printers, scanners, or digital cameras. There are several reasons to link a PDA through an infrared interface, including dialing from the directory or programming. Mitel Networks has introduced an IP telephone model with a docking station interface for a PDA. The PDA likely would function as the instrument’s display field, and provide data download capabilities for call processing and handling applications.

Ethernet power distribution. IP telephones, like PC clients, require power. Traditional PBXs power analog and digital telephones use internal power supplies to distribute power over inside telephony wiring. Converged (IP-enabled circuit switched) IP-PBX cannot distribute power across integrated IP gateway circuit cards to the LAN; neither can the LAN-connected telephony servers used in client/server IP-PBX designs. The first generation IP telephones were powered with an AC/DC transformer connected to a local AC power outlet. Each IP telephone required its own transformer and a dedicated UPS for emergency power support. Although an IEEE subcommittee had been working on its recommended standard for in-line power over an Ethernet LAN, IEEE 802.3af, Cisco could not wait and developed its own proprietary solution. Other proprietary solutions soon followed from other IP-PBX system suppliers, including 3Com and Alcatel. Third-party solutions, from suppliers such as PowerDsine, are available and work with IP telephones from other leading IP-PBX suppliers, such as Avaya and Siemens. In-line power options are currently priced at $50 to $100 per Ethernet port, but prices are expected to decline over time.

An Ethernet switch is equipped with an integrated or external power patch module, and power is distributed directly only to IP telephones, supported by the switch. Power is transmitted over unused Ethernet cabling wire pairs to only those Ethernet ports identifying themselves to the switch as IP telephone devices. IP telephones identify themselves to the LAN switch during an automatic self-discovery installation method or through manual programming by the system administrator. The Ethernet switch queries the IP telephone as to how much power is required or assumes a default power level.

Some of the basic specifications of IEEE 802.3af are:

  • DTE power shall use two-pair powering, where each wire in the pair is at the same nominal potential and the power supply potential is between the two pairs selected.

  • The power detection and power feed shall operate on the same set of pairs.

  • The DTE power maximum voltage shall not exceed the limits of SELV per IEC 950.

  • For DC systems, the minimum output voltage of the source equipment power supply shall be at least 40 V DC.

  • For DC systems, the source device shall be capable of supplying a minimum current of at least 300 mA per port.

  • The solution for DTE powering shall support mid-span insertion of the power source.

  • 802.3af systems shall distribute DC power.

Until the IEEE 802.3af standard is finalized, IP telephones will continue to be powered by available in-line power options or local AC power transformers. need for stand-alone telephony gateway equipment linking a traditional PBX system and an IP router. Calls placed from an IP telephone can be routed directly across a LAN and WAN without IP telephony servers.

Compressed voice. Traditional digital telephones are designed with codecs that digitize analog voice signals into digital format using 8-bit word encoding and 8-KHz sampling, resulting in 64-Kbps digital transmission over inside wiring and across the internal PBX switching network. IP telephones can compress voice signals for lower transmission rates and decreased bandwidth requirements. The most common digital encoding schemes currently used for voice transmission over Ethernet and IP WAN networks are G.711 (64 Kbps), G.723.1 (5.3 to 6.3 Kbps), and G.729/A (8 Kbps). G.711 is traditional PCM (no compression), but the two other codec specifications use compression algorithms. The total bandwidth used for voice transmission with IP transmission protocol is greater than the noted transmission rates; about 16 Kbps of additional transmission bandwidth is required because an IP destination address and overhead signaling bits are added to the voice datagram packets. Compressed voice transmission creates an overhead delay factor that may affect the quality of a conversation, but the trade-off is the potential for more efficient use of expensive off-premises network transmission resources. A T1 carrier circuit that typically supports a maximum of 24 voice-grade channels can support an equal or greater number of voice channels, with sufficient available bandwidth for concurrent data communications transmission, if voice is encoded using G.729/A compression. Using an IP telephone for voice compression eliminates the need for stand-alone telephony gateway equipment linking a traditional PBX system and an IP router. Calls placed from an IP telephone can be routed directly across a LAN and WAN without IP telephony servers.

Other IP telephone functions that reduce transmission bandwidth requirements are VAD and silence suppression. VAD detects voice communications signals entering the handset mouthpiece (microphone), and silence suppression signals the onset of “silent” voice transmission. A telephone call usually has a high percentage of silence during a conversation between parties, often as much as 50 percent of total talk time. A circuit switched connection is highly inefficient because much of the time there is no voice activity, but 8-bit words of “silence” are transmitted. With VAD and silence suppression, an IP telephone can reduce bandwidth transmission requirements because packets are not continually transmitted when no one is talking. When there are no voice communications signals picked up by the IP telephone microphone, a special signaling packet is transmitted to the destination IP address indicating the beginning of a silent period, when no new voice packets are being transmitted between the two endpoints. When voice activity resumes, another signaling packet is forwarded to inform the destination IP address that incoming voice packets are now on their way, effectively ending the period of silence. VAD and silence suppression packets are transmitted only when someone is actually talking, resulting in fewer packets and more efficient use of network resources.

Web browser. The most significant feature difference between a legacy digital telephone and an IP telephone is the integration of an embedded Web browser and pixel-based display monitor. The first question most people ask about Web-enabled IP telephones is: “Why do I need a telephone with Internet access if I have a PC?” The manufacturers of Web-enabled IP telephones are quick to point out that their product should not be considered a replacement for a fully functional PC client, but as a supplemental communications device for access to information when data processing is not required. These new IP telephones are best described as network communications portals that combine telephony functions with access to network information servers.

Thin client IP telephones have many of the internal design attributes of a computer: CPU, memory, operating system, applications software, and embedded communications protocol stacks. The RTOS of the thin client IP telephone may be proprietary, as in the Cisco Systems 7940/7960 models or the popular VX Works RTOS used by the Siemens optiPoint 600. Avaya’s 4630 IP telephone was the first Web browser model with a color display and touch screen control. The use of color can greatly enhance the functionality and ergonomics of the desktop instrument, particularly when displaying graphic information or photographs. Touch screen control, instead of cursor control buttons, provides point-and-click mouselike activation of features and menu selection. A telephone with touch screen control is not new; industry veterans may recall the Northern Telecom M3000 digital telephone introduced in 1985.

General desktop applications using an integrated Web browser include:

  • Access to directories external to the IP-PBX system directory database

  • Messaging (voice, text, fax)

  • Web page information screens

  • Personal calendar

  • Conference planning

  • Transportation schedules and reservations

  • Financial data (real-time stock quotes, investor information)

The accompany diagram of the Avaya 4630 IP telephone with a color touchscreen display illustrates the various applications supported by an IP telephone with an integrated Web browser interface.

Figure 1: Screenphone applications.

Figure 2: IP screenphone applications.

Figure 3: IP screenphone applications.

Figure 4: IP screenphone applications.

Using a telephone for e-mail or calendar access may seem strange if a personal computer is only inches away on the desktop, but it can be quicker and easier with the telephone. Telephones are always “on,” and information access is immediately available at a touch of a button. Booting up a desktop computer is getting longer and longer, as each release of Windows becomes more and more complex and the number of programs loading grows even larger. Many companies have several antivirus programs that run a series of system and memory checks before the computer is ready for use. The reliability level of a telephone has proved to be at least an order of magnitude greater than desktop computers, and it is less likely that the telephone will freeze due to program interactions or some other operating system glitch.

The Web browser feature can be especially useful in vertical markets where voice station users do not normally have a desktop computer. The healthcare, retail, and hospitality sectors are characterized by a significant number of stations users who have voice-only instruments at their disposal. For example, many nursing stations still have dumb CRT terminals for information access. In the retail sector, most point-of-sale (POS) terminals have no Web server access. In hotels, guest rooms have telephones, and Ethernet ports, but no computers. There is also a sizable number of installed telephones across all industry sectors with no nearby PC client. Many telephones are not located on a desktop shared by a computer: lobby telephones; cubicle telephones; conference room telephones; and wall-mounted telephones in hallways, cafeterias, or locker rooms. An IP telephone with a Web browser can be used as an information kiosk in public locations, such as shopping malls, bus terminals, or airports.

Mobile. There are three subcategories of mobile telephones for use behind a PBX system: cordless, premises wireless, and cellular. PBX cordless telephones can be proprietary or standard 2500-type analog. Proprietary cordless telephones are supported by proprietary PBX port circuit cards and have a unique signaling and control channel that allows for multiple line appearances and full PBX feature access and performance (including display-based information). Usually using spread spectrum technology and operating in the 900-MHz frequency range, a proprietary cordless telephone can often be used as a substitute for desktop models. A growing number of circuit switched PBX systems supports this option, including Avaya, Nortel, Siemens, NEC, and Toshiba. Analog cordless telephones, the same type commonly used for residential applications, appear to the PBX system as 2500-type telephones and offer limited feature/function access but a degree of station user mobility not offered by fully wired desktop models.

Premises wireless handsets are included as part of a premises wireless telephony option working behind the PBX system. The wireless handsets for these systems are proprietary to each system’s controller cards and base station transceivers. Base station coverage is limited in terms of geography and traffic handling. Most base stations support radio transmission ranges of about 50 to 150 meters, and between 2 and 12 simultaneous conversations per coverage cell. The wireless handsets closely resemble consumer cellular telephones, with several notable differences. Several manufacturers market wireless handsets with multiple line appearance buttons, fixed and programmable feature/function keys, and multiline displays that provide station users with information and data comparable to those of desktop digital telephone models. The high cost of a premises wireless handset and the infrastructure required to support coverage and traffic has limited the appeal of wireless telephony options, despite the ability of the station user to stay in touch with the PBX system regardless of location within the customer premises.

The first generation of premises wireless handsets was based on traditional circuit switching TDM/PCM standards. The recent introduction of wireless IP telephony solutions allows customers to use the existing LAN infrastructure to support distributed base stations. IP-PBX systems can interface directly to the wireless LAN infrastructure, but an MG is required for work behind a circuit switched PBX system. A leader in wireless IP is Symbol Technologies, whose Spectrum 24 wireless LAN system supports a wireless IP handset for use behind a PBX system. The Spectrum 24 uses spread spectrum frequency hopping within the 2.4- to 2.5-KHz band for transmission between access point transceivers and handheld communications devices. Data rates up to 2 Mbps per channel are supported. Each access point serves as an Ethernet bridge and can support wireless transmission coverage up to 2,000 feet in open environments and up to 180 to 250 feet in a typical office or retail store environment. Symbol’s NetVision Phone system provides enterprise voice communications capability and allows for integration into an existing PBX system (via a gateway) for premises and off-premises communications. The system includes NetVision Phones, access points, and a telecom gateway (third party). Each access point typically can support between 12 and 16 active clients and up to 10 voice-only conversations. There is a voice prioritization algorithm at the access point and client levels to minimize voice transmission delays. Fast roaming and load balancing support hand-offs between access points. Access point pinging detects and tracks station devices. The NetVision Phone is based on the ITU H.323 standard and converts analog voice signals into compressed digital packets (G.729/A 8-bit sampling rate, 160 bytes per packet) that are sent via the TCP/IP protocol over standard data LAN networks with the CSMA/CA wireless access protocol. TCP/IP addressing is used to tie to an extension number or a name directory. Several dialing mechanisms are supported:

  • Direct entry of complete or partial IP addresses

  • Direct entry of an “extension” number

  • Speed dial operation via speed dial keys

  • Recall/redial of a previous number

  • Using a name directory internally mapped to an IP address

  • Pressing the Send button begins the keypad dialing process

NetVision is a single line telephone, with a second “virtual line appearance” to support two concurrent conversations (one line is active and the other is in the hold mode). Intercom calls are supported between the phones over the LAN infrastructure, including broadcast capability to any number of phones. A multiline display field provides for incoming CLID services, and fixed function keys are used for one-button feature access. Symbol also offers a NetVision Dataphone for use with Spectrum 24. This telephone handset has an integrated Web client for accessing applications and databases and bar code scanning capability. Proprietary versions of NetVision telephones are used by Nortel Networks and Mitel Networks behind their IP-PBX systems. The NetVision IP wireless telephony system interfaces to the IP-PBXs via port interface gateway line cards. The accompany diagram illustrates the integration of the wireless NetVision handsets into an Ericsson MD-110 PBX configuration (Figure 5). The NetVision terminals are typical of IP wireless handsets that are designed for enterprise mobile applications.

Figure 5: Virtual IP telephony extensions.

Premises cellular is the third mobility communications option. The same cellular handset used with network cellular services, such as Sprint PCS, AT&T Wireless, and Cingular, can also interwork with a PBX system for premises mobile communications requirements. The first premises cellular options required an on-site mobility server and cell transceiver that linked to a local carrier’s network. The mobility server provided an interface between the PBX system and the premises cellular infrastructure to support control signaling and feature support to cellular handsets while the station user was on the customer premises. This mobile communications option had several drawbacks, including cost (mobility servers and transceivers are expensive for limited numbers of subscribers) and network compatibility. The premises transceiver could link to only one cellular carrier service, such as TDMA or GSM. All premises subscribers required a cellular handset that worked with the same network carrier service. Although some business customers supplied their employees with a cellular handset and had a low-cost contract with a single service provider, the more likely scenario was that PBX station users had a great variety of cellular handsets supported by different network service carriers. A better solution was needed than an expensive cellular infrastructure linked to a single service provider.

Ericsson, a leader in mobile communications networks, developed a more cost-effective and flexible premises cellular option. The MD-110 Mobility Extension option is based on an integrated interface circuit card housed in the PBX’s port carrier that can support a cellular handset with the use of any type of service standard from any local carrier. An ISDN PRI trunk circuit link is used to network the PBX system to the cellular network. Dialing procedures from the cellular handset will be in line with the terminal’s existing network service procedures, plus fully support the MD110-procedures, including station features (via voice prompts) and network call routing. The Ericsson Mobility Extension option is carrier service provider and transmission/encoding independent.

PC client softphone. The final category of PBX telephones is the PC client softphone. There are several categories of softphones. The first generation of softphones was based on CTI desktop applications using first-party (desktop telephone API link) or third-party (client/server configuration) call control. The CTI-based softphone requires a telephone instrument (analog or digital) for voice transmission to/from the desktop. An IP softphone is a PC client functioning as the voice terminal using an integrated microphone/speaker option to support LAN-based voice transmission, with signaling and control to and from a telephony server over the LAN/WAN infrastructure. For implementation of either softphone, a station user accesses and implements PBX features (dialing, call answering, call coverage, call processing) using a keyboard and/or mouse control for a GUI computer screen. Communications solutions using PC client software tools offer station users many advantages over traditional telephone instruments, with a limited number of feature/line keys and relatively small noncolor display fields. The accompanying diagram is an illustration of the Nortel Networks i2050 soft client phone (Figure 6). Some suppliers also offer customized client keyboards with integrated handsets for use as a softphone. The accompanying photograph is a Siemens optiKeyboard designed for use with its family of softphone client solutions (Figure 7)

Figure 6: IP softphone: Nortel Networks i2050.

Figure 7: Siemens optiKeyboard.

Market demand for CTI-based softphones has been very weak. Many station users prefer to depress traditional telephone buttons to access features rather than interact with a GUI-based computer screen to perform drag, point, and click operations. Telephone instruments also offer a far greater degree of reliability than PC hardware/software and are not affected to the same level as AC-powered desktop computers by local power problems. A major problem associated with first-party control CTI softphones was the requirement of a relatively expensive digital telephone equipped with an API link to the desktop computer. Third-party control client/server CTI configurations could be implemented with a lower-priced analog telephone, but station user functionality is severely affected when the desktop computer fails or is not performing properly. The primary market for desktop CTI has been among call center customers because the current ACD agent position depends heavily on desktop computer equipment and GUI-based interactions, and the cost of the solution is not significant compared with overall contact center expenses.

The emergence of IP-PBX systems may spur demand for PC client softphones because the cost of the solution may be far less than that of a high performance IP telephone. There likely will be great resistance to IP softphones from most station users who have grown comfortable with traditional telephone instruments, but the many potential benefits of the new solution may stimulate market demand.


IP Telephone Design Basics

All manufacturers base their IP telephone on proprietary design schematics and circuitry, but there are common design elements across the unique terminals. IP telephone basics include:
  • User interface

  • Voice interface

  • Network interface

  • Processor complex and associated logic

The accompanying diagram illustrates the internal design elements of an IP telephone instrument (Figure 1)

Figure 1: IP telephone design.

The user interface elements provide four classic telephone user function interfaces: keypad for dialing numbers; a variety of keys for line and feature access; a display for user prompts, caller feedback, messages, and other call processing information; serial interface to allow communications to an external device, such as a PDA, to allow synchronization of telephone information; speed dialing; and customer programming. An audible indicator (ringer) is also included to announce incoming calls.

The voice interface converts input analog voice signals into 8-bit digital word bit samples. Speech signals are sampled at an 8-KHz rate to create a 64-Kbps digital bit stream to the processor by using a standard PCM codec. Voice signal compression and IP encoding functions are performed by processor complex elements. The processor complex performs voice processing, call processing, protocol processing, and network management software functions. The complex consists of a DSP for voice-related functions and a MCU for the remaining control and management functions. The DSP and MCU each have associated memory. DSP memory usually includes RAM and ROM elements; MCU memory usually includes RAM and Flash elements. The Flash memory element supports software upgrades.

The network interface allows the transmission and reception of voice packets to and from the telephone terminal based on 10BaseT or 10/100BaseT Ethernet running TCP/IP protocols. Some IP telephones may be equipped with multiple RJ-45 Ethernet connector ports and an integrated Ethernet hub/switch to support connections to the customer premises LAN and desktop PC clients. Newer IP telephones also may be designed with a USB connector port.

Basic IP telephone software modules include a variety of user interface drivers (display, keypad, ringer, user procedures), voice processing modules, telephony signaling gateway modules, network management modules, and system service modules. The voice processing software modules include a PCM interface unit; a tone generator (call progress tones, in-band DTMF signaling digits); a line echo canceler unit (ITU G.168-compliant echo cancellation on sampled, full-duplex voice port signals); an acoustic echo canceler for terminals equipped with a speakerphone; VAD; voice codec unit (compression and packeting of the 64-Kbps digital stream received from the station user based on a variety of algorithms, such as G.711, G.723.1, G.729/A, etc.); packet playout unit (compensation for network delay, jitter, and packet loss); packet protocol encapsulation unit (based on RTP, which runs directly on top of the UDP); voice encryption (to ensure privacy); and a control unit (coordinates the exchange of monitor and control information between the voice processing module and the telephony signaling and network management modules).

The telephony signaling gateway subsystem in an IP telephone performs the basic functions for call setup and teardown procedures. Software modules used by this subsystem include call processing, address translation and parsing, and network signaling. The most widely implemented network signaling standard used by currently available IP-PBX systems is H.323 protocol. H.323 is an ITU standard that defines several signaling and protocol specifications for multimedia communications between LAN-based terminals and network equipment. The main H.323 standards used for VoIP in an IP telephone are H.225–Call Signaling Protocol (based on Q.931), H.245–Control Protocol; RAS Protocol; and RTCP. An emerging network signaling standard not currently used by any commercially available IP-PBX, but being planned for by most suppliers, is SIP. SIP is the protocol developed and promoted by the Internet Engineering Task Force (IETF) and is forecasted to be widely implemented in network hosted services, such as IP-Centrex, and may eventually replace H.323 as the primary signaling protocol used by premises communications systems.


Voice Terminal Categories

PBX voice terminals can be classified into several basic telephone categories:
  • 2500-Type analog telephone

  • Digital telephone

  • Mobile

  • PC client softphone

Within the latter three categories are several subcategories. The distinguishing technical difference between an analog telephone and a digital telephone is that the latter terminal has an integrated codec that digitizes voice signals for transmission to the PBX system over the inside wiring system. The digital transmission format can provide a higher degree of feature and function performance at the desktop level, instead of at the PBX system common equipment. Before the design, development, and widespread availability of digital telephones, the first generation of stored program control PBX systems supported electronic telephones, sometimes referred to as hybrid telephones. Voice transmission between the desktop and the PBX system was analog, but the telephone design included integrated circuitry, sometimes a microprocessor chip, that could support multiple line appearances. Programmable feature/function buttons and limited function display fields were supported, but the performance capabilities were limited due to in-band signaling techniques between the port circuit card and the desktop. Digital telephones, supported by an out-of-band signaling channel, were capable of far greater performance potential.

Mobile telephones is a term used to categorize cordless, wireless, and cellular telephone handsets. Cordless telephones working behind a PBX system are likely to use digital radio transmission between the handset and the base station, but analog voice transmission is supported between the base station and the PBX port circuit card. The terms wireless and cellular usually describe telephones that do not require any local loop wiring, although the base station transceiver is hardwired back to a central switching system, such as a PBX system.

PC client softphones are based on a CTI platform from which the PBX common control complex functions as a server to the desktop terminal. With a client softphone option, communications control and signaling between the desktop and the PBX system is handled over a CTI link (desktop or client/server configuration) behind a traditional common control complex or LAN-based call server. The former type of softphone application requires an associated telephone instrument for voice communications transmission at the desktop. The latter is an example of a softphone based on an IP telephony platform, without a traditional desktop telephone. An IP-based softphone requires the PC client be equipped with a sound board with a combination microphone/speaker or a computer handset.

Analog Telephones

DTMF analog telephones that conform to the old Bell System 2500-type standard specifications are nonproprietary and can be supported by all PBX systems. However, the analog port circuit cards for each PBX system are proprietary. Transmission between the telephone and the PBX is analog based, with in-band signaling for all dialing and feature activation operations. Analog-based voice communications and signaling, transmitted over a 4-KHz transmission line, is carried over two-wire (single pair) UTP telephone wiring between the wall jack and the PBX system. The embedded signaling bandwidth limits support of integrated desktop features and functions, particularly display-based information. Analog telephones typically have a single line appearance, although a second virtual line may be supported for answering incoming calls after the original connection is placed on hold. Two line appearance analog telephones require multipair wiring and multiple port circuit card terminations. Analog telephone in-band signaling over a single wiring pair does not support multiple line appearances behind a PBX system.

Analog telephones may be equipped with a limited number of fixed feature buttons, such as hold, and an array of programmable speed dial buttons. The instrument may be equipped with a message indicator and limited function display field. Displays are usually limited to dialing and call duration information, time clock, and caller line ID (CLID) incoming call directory numbers. More detailed information such as name display, call diversion information, or feature/function menus commonly available in more sophisticated, and more costly, digital telephones are not available.

Some analog telephones have an integrated hands-free answer intercom speaker or a two-way simplex speakerphone. Almost all currently available desktop audioconferencing products are based on standard analog transmission standards between the desktop and the PBX system and supported with the same analog station port circuit used for 2500-type analog telephones. Audiconferencing products are typically equipped with a DTMF keypad and several fixed feature buttons, e.g., mute, and support full duplex speakerphone operation. The Polycom Soundstation is an example of an analog-based desktop audioconferencing product.

Digital Telephones

The second category of voice terminals is digital telephones. Digital telephones have a codec that digitizes analog voice signals at the desktop, using PCM as the encoding scheme. Excluding IP telephones, digital telephones are supported through an out-of-band signaling and control channel between the desktop and PBX port circuit card. There are several subcategories of digital telephones: PBX systems, it is used to support advanced features unique to each system, particularly display-based system capabilities and functions. Figure 1 illustrates a typical multiple line digital telephone instrument from Avaya.

Figure 1: Typical multiple line appearance digital telephone.
  • Proprietary

  • Universal Serial Bus (USB)


  • IP

Proprietary. Most digital telephones currently working behind a PBX system are proprietary, and work exclusively with the manufacturer’s PBX system(s). Some manufacturers, such as Siemens and NEC, have designed their digital telephones to work across different product families of communications system (KTS/Hybrid and PBX systems). All current proprietary digital telephones are supported by a 2B+D communications/signaling transmission format between the desktop and port circuit card. Although the 2B+D transmission format is most closely associated with ISDN BRI services, the dual communication channel/dedicated signaling channel format was first implemented on a proprietary digital PBX telephone in 1980, before there were ISDN standards. The first digital PBX telephone was the Intecom ITE model, capable of supporting digital communications from desktop to desktop across the PBX cabling infrastructure and switching network, with an optional data module for supporting modemless data communications from the desktop. Each B channel can support 64-Kbps communications transmission for voice, data, or video signals. What makes a digital telephone proprietary is the D-channel signaling protocol. The protocol is proprietary and unique for each manufacturer’s PBX system. Although the proprietary nature of the digital telephone’s D-channel format restricts customer flexibility in selecting telephones for use behind their PBX systems, it is used to support advanced features unique to each system, particularly display-based system capabilities and functions. Figure 2 illustrates a typical multiple line digital telephone instrument from Avaya.

USB. A special type of proprietary digital telephone is uses an integrated Universal Serial Bus (USB) interface. USB is an external bus standard capable of very high-speed transmission rates, up to 12 Mbps, and can support a wide variety of communications applications. The original intent of a digital telephone equipped with a USB interface port was for desktop CTI applications. The USB port passes signaling and control messages/commands between the voice terminal and a desktop computer. Each PBX manufacturer originally designed proprietary CTI API port interfaces for implementing desktop PC telephony applications. It was believed that the design, development, and adaptation of USB standards would stimulate the market for desktop CTI installations behind PBX systems, but few manufacturers incorporated the interface port in their telephone instrument designs.

A recent innovation using the USB link supports IP telephony. A digital telephone with a USB port eliminates the need for a RJ-11 jack interface between the phone and the PBX cabling system because communications transmission and signaling can be handled over the LAN infrastructure by using a desktop computer as the intermediary link to the LAN. The failure of CTI as a PBX station option has historically limited demand for USB telephones, but the emergence of ToIP communications may help create future demand for the option. Customers using a USB telephone in a ToIP installation can continue to enjoy the look and feel of a traditional telephone with a traditional keypad and handset for dialing and call answering operations while using the desktop computer to facilitate feature/function access (using CTI client software). Frequent computer processing and software problems, a major reason most station users are reluctant to use a softphone, do not affect most USB telephone functions or operations because the computer serves only as a physical connection to the LAN. Of all the major PBX system manufacturers, Nortel Networks has been the most active in promoting use of its USB telephone model for ToIP applications behind its IP-PBX communications systems. pling before a VoIP audio codec converts the digitized sample to IP format. IP telephones may conform to VoIP protocol and signaling formats standards, but each IP-PBX system uses proprietary signaling bit data in support of unique features and display characteristics. The accompanying photograph of the Cisco 7960 illustrates a current generation IP telephone (Figure 2). Although most IP telephones with Web browser capabilities have similar feature capabilities, the look and feel of each supplier’s telephone models is distinct. The accompanying photographs of three supplier’s high-end IP phone models illustrates this (Figure 3).

Figure 2: IP phone attributes: Cisco 7960.

Figure 3: Contrast in styles: IP telephones with Web browser displays.

ISDN BRI. A third category of digital telephones is ISDN BRI. The ISDN BRI telephone communications link to the desktop is 2B+D, but unlike proprietary sets, its D-channel signaling format conforms to National ISDN (NISDN) specifications. ISDN BRI telephones have limited access to some proprietary PBX features, and the level of display information is also limited compared with proprietary digital telephones. Although ISDN BRI telephones have been available since the early 1990s, support of ISDN BRI telephones offers customers some benefits not available with proprietary digital instruments, such as passive bus operation and bonding of the two communications bearer channels. The former is the ability for a single ISDN BRI 2B+D communications link to support up to eight desktop telephones, each with its own directory number. Although only two telephones can be active simultaneously, customers can save money on port circuit hardware and cabling when these telephones are used in low-traffic environments. The latter application supports high-speed transmission to the desktop, up to 128 Kbps, for data or H.320 video communications applications. pling before a VoIP audio codec converts the digitized sample to IP format. IP telephones may conform to VoIP protocol and signaling formats standards, but each IP-PBX system uses proprietary signaling bit data in support of unique features and display characteristics. The accompanying photograph of the Cisco 7960 illustrates a current generation IP telephone (Figure 2). Although most IP telephones with Web browser capabilities have similar feature capabilities, the look and feel of each supplier’s telephone models is distinct. The accompanying photographs of three supplier’s high-end IP phone models illustrates this (Figure 3).

IP. The fourth digital telephone category, and the most recent, is an IP telephone. IP telephones are included in the digital category because voice signals are digitized with standard 8-bit coding and 8-KHz sam- pling before a VoIP audio codec converts the digitized sample to IP format. IP telephones may conform to VoIP protocol and signaling formats standards, but each IP-PBX system uses proprietary signaling bit data in support of unique features and display characteristics. The accompanying photograph of the Cisco 7960 illustrates a current generation IP telephone (Figure 2). Although most IP telephones with Web browser capabilities have similar feature capabilities, the look and feel of each supplier’s telephone models is distinct. The accompanying photographs of three supplier’s high-end IP phone models illustrates this (Figure 3).


Overview of ANSI/TIA/EIA 569

ANSI/TIA/EIA 569 is the Commercial Building Standard for Telecommunications Pathways and Spaces. The purpose of 569 is to standardize design and construction practices within and between buildings that support telecommunications equipment and transmission media. The standards are outlined for rooms, areas, and pathways into and through which telecommunications transmission media and equipment are installed. The standard is limited to the telecommunications aspect of building construction and design and does not cover safety aspects.

The specifications of 569 cover the following building elements:

  • Entrance facilities

  • Equipment room

  • Backbone pathways

  • Telecommunications closet

  • Horizontal pathways

  • Workstation

The entrance facility, equipment room, telecommunications closet, and workstation areas were described briefly in the preceding section on 568. The backbone and horizontal pathways are used for the corresponding cabling described above. Backbone pathways consist of intra- and inter-building pathways. Intrabuilding pathways consist of conduits, sleeves, and trays. They provide the means for routing cables from the entrance facility to telecommunications closets and from equipment rooms to the entrance facility or the telecommunications closet. Interbuilding pathways interconnect separate buildings and consist of underground, buried, aerial, and tunnel pathways. Horizontal pathways are facilities for the installation of the telecommunications transmission media from the telecommunications closet to the telecommunications outlet at the workstation area.

The 569 specifications require a minimum of one telecommunications closet per floor, and that additional closets should be added if the floor area to be served exceeds 1,000 square meters or the horizontal distance to the work area is greater than 300 feet. At least one telecommunications outlet per workstation area is specified.

A very important area covered by 569 is labeling and color-coding specifications designed to simplify installation and maintenance of the cabling infrastructure.

Labels are divided into three categories: adhesive, insert, and other. Adhesive labels must meet UL requirements for adhesion, defacement, legibility, and exposure. Insert labels must meet UL requirements for defacement, legibility, and exposure. Other labels include special-purpose labels, such as tie-on labels.

The 569 color coding rules are:

  • Termination labels at the two ends of the cable should have the same color

  • Crossconnections between termination fields generally should have two different colors

  • The color orange is used for the demarcation point

  • Green identifies network connections on the customer side of the demarcation point

  • Purple identifies the termination of cables originating from common equipment

  • White indicates the first level of the backbone media

  • Gray indicates the second level of the backbone media

  • Blue identifies the termination of station telecommunications media

  • Brown identifies interbuilding backbone cable terminations

  • Yellow identifies the termination of auxiliary circuits, alarms, security, and other miscellaneous circuits

  • Red identifies the termination of KTSs

  • White may be used to identify second-level backbone terminations in remote “non-hub” buildings

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