Networking

Review: Polycom SoundStation Duo

This conference room phone provides multiple communications options, directory integration and remote management for complete enterprise integration.

Wayne Rash

Wayne Rash is a longtime technology journalist who has directed product testing centers. He is President of Wayne Rash & Associates, an analysis and editorial services firm located near Washington, D.C. He can be reached at wrash@mindspring.com. You can also follow him on Twitter as @wrash.

The Polycom SoundStation Duo incorporates nearly every communications function an organization might expect from a conference phone, as well as a few they might not realize they need.

Besides producing high-quality voice, the SoundStation Duo operates as a full-duplex phone, which means that the parties on either end of the line can hear each other even when they talk simultaneously — a rare feature in speakerphones. It’s also a dual-mode phone that works as a Voice over IP (VoIP) device and as an analog phone running on the public switched telephone network (PSTN) or an analog private branch exchange (PBX). Equally rare, it can perform these functions simultaneously. What’s more, the SoundStation Duo will connect with many Android and iOS smartphones.

Organizations should note that the phone’s VoIP features require a Session Initiation Protocol–capable network. Those lacking a SIP–capable network can still use the SoundStation Duo with their analog phone system, and may be able to use their existing directory service for calling information. The phone works with Microsoft Active Directory and most Lightweight Directory Access Protocol (LDAP) directories, and it also supports Microsoft Lync.

End-User Advantages

Complex infrastructure notwithstanding, the SoundStation Duo is extremely easy to use. Phone lists consist of numbers from the network directory, numbers that have been called from the device and numbers that have called into the device. To place a call, the user can either select a name from the list that appears on the phone’s LCD screen or punch in a number using the keypad.

Voices were clear on both ends, and the phone’s extension microphones delivered quality audio even when tested in a fairly large conference room. The sound quality is better in VoIP mode than when using the analog PSTN, but even with traditional phone service, the quality was quite good. Users also will like having the ability to connect smartphones, tablets and computers to the phone (although a special cable, available through Polycom, is required to do so).

Why It Works for IT

The SoundStation Duo is designed specifically for the enterprise. In addition to supporting an organization’s LDAP implementation, the phone has a built-in, web-based configuration utility that requires just six steps. To manage the phone remotely, a network administrator simply needs to browse to the IP address, which can be found on the phone’s LCD panel, and enter the required password. The web-based setup and configuration process is intuitive and accessible, and includes a context-sensitive help column on the right side of the screen that explains how each function is used and provides examples of potential entries.

3
The number of cardioid microphones in the SoundStation Duo

Polycom has designed the SoundStation Duo to work with most IP PBX devices and servers, and it takes advantage of most SIP features. However, unlike many SIP phones, this one doesn’t require a remote boot server. It’s worth noting that the SoundStation Duo supports automatic failover to the analog line, should the VoIP communications fail. The phone also will use either the SIP or the analog line to make and receive calls.

The SoundStation Duo supports, but doesn’t require, Power over Ethernet. It also supports advanced PBX features, such as one-button conferencing, and sets its time and date using external timeservers. The device works with most SIP servers and supports mass provisioning.

Disadvantages

If calls will be placed outside the organization, the firewall must be provisioned to allow for SIP packet pass-through. Those using SIP likely will have already completed this task, however.