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:
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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.
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The power detection and power feed shall operate on the same set of pairs.
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The DTE power maximum voltage shall not exceed the limits of SELV per IEC 950.
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For DC systems, the minimum output voltage of the source equipment power supply shall be at least 40 V DC.
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For DC systems, the source device shall be capable of supplying a minimum current of at least 300 mA per port.
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The solution for DTE powering shall support mid-span insertion of the power source.
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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:
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Access to directories external to the IP-PBX system directory database
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Messaging (voice, text, fax)
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Web page information screens
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Personal calendar
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Conference planning
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Transportation schedules and reservations
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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.
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:
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Direct entry of complete or partial IP addresses
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Direct entry of an “extension” number
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Speed dial operation via speed dial keys
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Recall/redial of a previous number
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Using a name directory internally mapped to an IP address
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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.
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)
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.
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