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title IPW-2000 Won the Editor's Choice Award from the Industry Leading Internet Telephony Magazine date 2009. 8. 31

The documentation consists of a Quick Start Guide and a User License Agreement for the gatekeeper software. The first half of the Quick Start Guide gives instructions for the basic setup and configuration scheme of the IP Centrex phone. The second half explains the basic features, some user information, and some troubleshooting issues for the phone. While this guide is not flashy and has grammatical errors throughout, it does cover most of what a user might want to know about the phone, and provides many screen shots and other graphics to illustrate the information more clearly.

The primary feature of the IP Centrex phone is for placing VoIP calls over the LAN from one IP Centrex phone to the other. The phone is fully compliant with H.323 version 2 and supports the G.711, G.723.1, and G.729 compression codecs. The use of enhanced jittering compensation and bad frame interpolation aid call quality.

The rest of the features of the IP Centrex phone, such as an LCD screen, speakerphone, speakerphone mute, microphone and speaker ports for a headphone, redial, ringer change/volume adjustment, speed dials, and caller ID, are on most basic digital telephones. Future enhancements will include a digital answer machine with voice recording, Internet services such as a remote configuration manager, and a WML browser for accessing content. More PBX functionality (e.g., call forwarding, three-way call conferencing, and voice mail) is offered as a supplementary service, which is dependent on the capabilities of the switch.

Once we had the gatekeeper and both IP Centrex phones up and running, we first checked the network information to make sure it was correct, then dialed the four-digit number of the other IP Centrex phone, and followed by pressing the pound key. The other phone rang almost instantaneously, and we saw the packet information run through the gatekeeper.

Upon pickup, we examined the sound quality and latency of the connection. We could hear each person speaking very clearly with no echo and a minimal amount of delay. The quality was that of a regular PSTN call, not that of a VoIP call. As a matter of opinion, this particular VoIP call sounded better than many "regular" phone calls we made that day. When making additional VoIP calls using the IP Centrex phone, we had similarly excellent results.

As for the more fundamental telephony aspects of the IP Centrex phone, we were able to create a speed dial for the first available button and make a call to the other phone by pressing either of two buttons: F1 or the second button on the upper left side of the phone (pressing the Menu button shows which button should be pressed for specific speed dials). We expect that the Menu button will offer more functionality in future versions of the IP Centrex phone.

We assumed that we could not test the flash button for transferring calls since we only possessed two IP Centrex phones. We did not have a third phone to register with the gatekeeper, and pressing the flash key actually disconnected the current call.

We quickly realized that if we registered a NetMeeting client to the gatekeeper, we might be able to transfer or conference a call. While we were able to register a NetMeeting client and make one-to-one calls through the gatekeeper from NetMeeting to one of the IP Centrex phones, the connection still hung up when trying a transfer or a conference call. With the addition of this third registered connection, we became relatively sure that this was because of the limited functionality of the gatekeeper we were using. A more feature-rich gatekeeper would have allowed us to transfer or conference a call. This test, however, did prove the IP Centrex phone's H.323 compatibility with NetMeeting.

We think it might interest our readers to note that we first became aware of this product during ConvergeNET this past October at INTERNET TELEPHONY CONFERENCE & EXPO in San Diego. It was there that we originally performed some live demonstrations with the IP Centrex phone. Instead of the simple gatekeeper we used for the above tests, we used gatekeepers from Tundo and NetCentrex and connected to other H.323 compatible equipment. The voice quality was good doing these tests as well, although not quite as excellent as the lab tests. Just as important was that by performing these tests we proved that the IP Centrex phone was interoperable with other company's equipment.

While we have no complaints about the VoIP functionality of the IP Centrex phone, the suggestions we do have are negated when the product is integrated with AG Communications / Lucent Technologies' iMerge product. With this collaboration, we hope to see client software for the PC (either Web-based or Windows-based), which has right-click functionality and help files, and allows for remote management functionality as well as desktop call control. From the phone, it would also be beneficial to push Web pages, coupons, and other material from the Internet to the LCD screen. While this might already be in the works for UniData's WML browser, it is still worth mentioning.

In addition, we would recommend adding an analog port so that a fax or modem could connect to the IP Centrex phone. The port could act as a digital to analog converter to allow for such functionality as Fax over IP.

Calls can be made from an IP Centrex phone to a gatekeeper set up in an ISP, CLEC, or ITSP, and then either continued as an IP call or hopped-off to the PSTN and on to the "outside world." To achieve these ends, UniData's IP-Centrex phone is one of the first of its kind, and is a quality product, especially when it comes to its main purpose -- VoIP calls through a LAN. The voice quality is among the best we've heard, and installation is a snap. We award this product our highest honor for its ease of use and installation, and the sheer quality of the VoIP calls.

Source link: http://www.tmcnet.com/it/0301/0301labs.htm

l UniData's IP Phones Awarded CT Expo 2001 Best of Show
l Lucent Technologies to Present VoIP Interoperability At Internet Telephony Conference and Expo