Buying an IP Phone

Before you purchase a voice over IP phone system, you need to consider the form, location, interoperability and performance--or FLIP--of the phone. We'll help get your purchasing plans started.

May 6, 2005

4 Min Read
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If you have the budget, let users choose from a number of design options. If you must keep your costs down and buy in bulk, create a user committee and get buy-in from a representation of users.

Location, Location, Location

Location determines the function of IP phones and whether they need basic, executive or console attendant features (see "Distinguishing Features," page 78). So first ask: "Where will the phone be used?"

Basic IP phones are, well, basic. Most reside on employee desktops; some sit in common areas, such as lunchrooms, reception areas, mailrooms and loading docks. These phones should have at least a multiple-line LCD display to show caller ID and current-call status, or the current call and a call on hold. Hard-key functions for volume, hold and transfer should be obvious.Many basic phones also have message-waiting indicators for voicemail, adjustable screens with backlit displays, and at least five programmable keys for redialing and speed dialing. Both Snom Technology's Snom 190 and Siemens' optiPoint 410 are good basic IP phones at the midrange price of $200 to $400.

Power users, managers and executives will want the more sophisticated models--phones with more programmable function keys to transfer calls and conference other users, as well as additional line and bridge appearances. These phones might sport a full graphic display rather than a multiline LCD and have a built-in, full-duplex speaker, like the Polycom SoundPoint IP 500 does. The 500 has three call and bridged appearances, 13 programmable keys, a graphic display of 160x80 pixels and a full-duplex speaker. Cisco Systems' 7960G and ShoreTel's IP 560 are other phones in this class. High-end phones typically cost $400 and up.

Attendant phones are for console operators who need to set up conference bridges, route incoming calls and forward calls. Attendant phones need the most line and bridge appearances possible, so they're usually coupled with a vendor's soft-phone app to use a PC interface.

Soft phones increase the available programmable keys for speed dialing, let operators view an entire corporate directory at the click of a mouse, and let the operator drag and drop call appearances on user objects.

Many VoIP systems provide interoperability with standards-based IP phones and include support for H.323 and MGCP (Media Gateway Control Protocol). But the de facto signaling standard for the industry is SIP (Session Initiation Protocol), which lets endpoints easily communicate with each other without proprietary gateways or translation services. Alcatel, Nortel Networks and others are slowly but surely migrating to SIP, while Avaya, Vonexus and Zultys Technologies have fully embraced it.Using SIP phones will make it easy to integrate with other SIP-compliant devices like videoconferencing endpoints and apps, such as IM and presence management. For example, Nortel's phone system supports Polycom multimedia desktops and group conferencing systems. And Siemens' HiPath system uses Microsoft's Real-Time Communication Server for presence management.

Avaya's 4602SW phone is SIP-compliant and has all the features and functionality of a Cisco 7960G, yet at a better price. The 4602SW is also less costly than popular third-party SIP phones from Snom, yet it contains hard keys for volume as well as hold, transfer and forward. Polycom also embraces VoIP standards and includes support for H.323, MGCP and SIP in its IP phones.

Network Performance

VoIP systems are sensitive to network performance. On the one hand, VoIP increases network bandwidth requirements and may impact other enterprise applications over WAN links. On the other hand, a busy network with high latency or delay can result in choppy speech and reduce call quality below toll quality.

Most IP phones support standard codecs like the G.711, G.729 and G.723, which sample and compress speech for transit and decompress speech on receipt to save bandwidth. The G.711 codec uses PCM (pulse code modulation) to sample and transmit speech at approximately 80 Kbps over IP. If your bandwidth is limited, the G.729 codec compresses speech to approximately 40 Kbps but adds delay to process the compression algorithm. The G.723 codec further compresses the voice data but adds even more delay between endpoints. If you want to save bandwidth without increasing delay, make sure the codec you use supports VAD/SS (voice-activity detection with silence suppression). Silence suppression cuts down on bandwidth use by filtering out background noise and transmitting packets only when someone is talking.Final Factors

A few other features will facilitate administration. Make sure the phones are easy to upgrade and configure with a TFTP server. Other useful features to help manage the phones include call logs, diagnostic call-quality information and manual settings to adjust phone ports to half or full duplex as necessary. In some instances, an intermediary device like a switch or PC may not autosense the phone's connection and a manual adjustment is needed.

Furthermore, don't miss some of the IP phone innovations coming down the pike. For example, Mitel Networks' 5230 IP phone can fully integrate with PDAs and even act as a docking station for them. This will make it possible to activate phone features on PDAs and transfer phone capabilities from one device to another.

Sean Doherty is a senior technology editor and lawyer based at our Syracuse University Real-World Labs®. Write to him at [email protected].

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