Making IP Voice Calls

In a converged IP-PBX system, a call begins when the IP telephone goes off-hook and the PBX common control complex is alerted via a control signaling path between the IP telephone and the common control complex functioning as a gatekeeper. The signaling is transmitted to and from the LAN via an IP port interface card, dedicated Ethernet TCP/IP interface card, or integrated Ethernet connector. Dial tone packets or a control signal will alert the IP phone to activate a dial tone signal, an indication to the station user that the IP phone is ready for use. As the calling number digits are dialed, the common control complex will send multiple signals to the desktop, such as ring back or other call progress tones. If the call is being placed to a non-IP station port, such as an analog telephone, or a PSTN trunk circuit is required for an off-premises call, the following steps will occur:

1. A local TDM bus talk slot will be assigned

2. A gateway bearer communications channel will be assigned

3. Voice communication signals will be transmitted over the LAN to the IP port circuit card

4. The packeted voice signals will be reformatted, buffered, and transmitted across the gateway channels to the local TDM, and the call will continue until one party releases (disconnects) the connection.

For purposes of the call, the IP telephone appears to the common control complex and switching network as a PCM peripheral endpoint for voice transmission and feature activation operations. Converged IP-PBX systems typically use the same proprietary control signals for IP telephone support as those used for digital telephones.

If an IP station user is placing an intercom call to another IP telephone, the common control complex will direct the IP telephone to start sending voice packets directly to the IP address of the called IP telephone. The common control complex also will direct the called party’s IP telephone to send voice packets directly to the IP address of the calling party’s IP telephone. The direct audio communications path between the two IP telephones, using only LAN facilities, is often referred to as an IP-PBX peer-to-peer LAN switched connection or direct IP. No traditional PBX circuit switched connections are used for the call.

Direct IP connections will be established automatically between two IP endpoints if several conditions are satisfied:

• Both IP endpoints are administered to allow direct IP connection

• No TDM connections are required for either IP endpoint, and a point-to-point LAN connection is available

• The IP endpoints are in the same network region or in different network regions that are interconnected via LAN/WAN facilities

• The two IP endpoints share at least one common voice codec in their voice codec lists and the internetwork region connection management voice codec list

• The IP endpoints have at least one voice codec in common, as shown in their current codec negotiations between the endpoints of the IP-PBX

If any of these conditions are not satisfied, the call may require TDM connectivity.

A direct IP connection established for an existing call may be torn down if circuit switched TDM connectivity is required during the call. Conditions that may require TDM connectivity, based on the IP-PBX system, are:

• Additional parties are conferenced onto the call, including IP endpoints

• A PBX signaling tone or announcement needs to be inserted into the connection

• The connection is put on hold—music on hold

When the event requiring TDM connectivity is no longer in effect, a direct IP connection may again be established. The generic term for call connections that change from direct IP to TDM connectivity and back to direct IP is null capability.

The first generation of converged IP-PBXs required talk slots on the local TDM buses to support multiparty conference calls for calls among IP endpoints, even if all the parties were IP endpoints. Each conferenced party required a talk slot per TDM local bus to support the call. The manufacturers have future plans to support non-TDM conference calls among IP endpoints through enhanced versions of their current IP port circuit cards. Planned versions of IP port circuit cards will include conferencing circuits to support multiparty calls exclusively among IP endpoints, thereby eliminating the need for TDM switched connections. The IP endpoints may be internal IP telephones or IP trunk circuits connecting off-premises stations.

Examples of IP Phones

The large telecom equipment manufacturers, such as Lucent and Nortel, are addressing the IP phone market with just one or two models, compared with the wide variety of telephones that they previously made for their legacy Centrex and PBXs. Presumably these vendors hope to rationalize their production line and minimize the costs of making large quantities of a standard set. The first generation of IP phones from these large companies, including Cisco, were proprietary devices that could only be used with that vendor's switches. The trend now is to produce sets that are based on the SIP or Megaco standards, or both.

Other independent phone suppliers (such as Aastra, Avanti, Pingtel, Telcordia, and Telrad) are offering intelligent, voice-over-IP phones to complement the features of IP-Centrex. Well-established, high-volume handset producers, such as Samsung and Sony, are also supplying this market.

Two telecom manufacturers, Mite1 and Siemens, have earned a reputation for ergonomically excellent telephone terminals, and both produce a range of IP phones. Mitel has five IP phone models, including a low-cost, single-port unit and the 5140 IP Appliance, which is pictured in Figure 3. The more expensive device has a 320-by-240 pixel display and an infrared adapter (IrDA) interface, which provides a link to (personal digital assistants) PDAs. This phone enables users to define icons to represent telephony features, using the Palm-based graphical interface.


Figure 3: Mitel's 5140 IP Appliance. (Reproduced with the permission of Mitel Networks.)

The Mitel 5140 also has a built-in Hypertext Markup Language (HTML) browser and integrated directory management capabilities. It can be used as an agent's or supervisor's workstation in an automatic call distribution (ACD) configuration.

An interesting and forward-looking type of IP phone is one that is essentially a PDA cradle, built to accommodate a Palm Pilot or a similar device. The PDA provides a color screen display and most of the processing power, as well as its built-in operating system, applications, and database. This keeps the cost of the phone itself very low and provides a powerful desktop terminal.

The Siemens IP phone brand is optiPoint; these were the first sets from a major telecom manufacturer to support SIP. The optiPoint 100, for example, has both 10 Mbps Ethernet and RJ-45 interfaces and includes the G.711 (64 Kbps) and G.723.1 (5.3 Kbps) voice encoding algorithms. It has a two-line, 24-character display and is a hands-free, speakerphone set.

A major advantage with IP phones is that, when moved, these phones automatically reregister with the communication system, providing access to the voice services within a few seconds. Most IP phones also integrate with HTTP-based system management packages to allow for fast and intuitive moves, adds, and changes (MAC).