Ability to accept and place IP Phone calls
codec support for G.711
Support for at least one of the following protocols: H.323, SIP, or MGCP

The product may optionally include one or more of the following features:

Voice packet QOS support
Speakerphone or hands-free capability
Data port connectivity (via 1 or more data Ethernet connectors, with or without a true switching capability)
Alternate codec support for G729A or AB, G.723.1, etc.

An overview to the multivendor comparison as well as links to the other vendor reviews can be found here.

Each vendor was judged and reviewed on:

Ease of Installation;
Product Features;
Product Documentation/Integrated Help;
Technical Support;
the Results of Manual Phone Exercises/Reviews;
and the Results of Automated Speech Quality Tests.

For more on the automated test procedures, including notes on how CT Labs did it and details on the SMOS and PVIT tests, see .

Install of the optiPoint 600

optiPoint 600 wasn't the easiest phone to initially work with.

Attaching the key module was very easy, even though there was no "Quick Start" type of instructions. But lack of simple instructions did make it hard to find out how to put the phones into peer-to-peer mode with static IP addresses. It was also hard to figure out how to find the configuration menu (use the stylus to press on the tiny triangle right next to "Personal" on the screen) and the main phone menus (push the right scroll key).

Since almost all the function keys are programmable, we had to program all the functions that we wanted to test into buttons before we knew we could test them from the phones. According to Siemens, this is only when they are in peer-to-peer mode, which is not a normal configuration for anyone who buys their phones.

Normally, they are put into use with a server configuration, where the setup is easier (due to DHCP, etc.), and you can load configuration files to the phones instead of programming each button manually.

Product Features

The phones had a high-quality look and feel. The buttons weren't labeled (mainly because most of them are programmable) and it's not obvious how to find the menu or configuration interfaces. The buttons were also a little small and close together " easier to "fat-finger." Not horrible, though.

The menus are hard to find " one menu is accessed by hitting the right-arrow button, and then the menu text will show on the bottom line of the display. To maneuver through the menu, you continue to hit the right-arrow key to see one choice at a time. When you see the choice you want, you hit the check-mark key. The other menu is accessed by using the stylus to press a small triangle on the screen that looks like it's part of the "personal" selection. From there, the menu choices show as a list, displaying on the whole screen.

Since everything is programmable, you can make custom feature buttons for just about anything.

We tried out an external loudspeaker/microphone unit that turns any optiPoint 600 phone into a conference-type phone " very cool! We also tested a "sidecar" unit " an additional keypad that can be added to the phone to allow more programmable keys.

The display is backlit, which helps viewing, and you can tilt it. will accept stylus input (very cool) for certain menus and applications that can run on the phone. The phone will run Java applications that can directly use the phone display. The display is 320 x 240 pixels (8 x 24 characters).

Besides the dialpad, there are 2 volume keys, 3 navigation keys, and 19 programmable keys/status LEDs. Snap-in key module adds 16 programmable keys. Programmable keys can be feature keys or speed dials.

Programmable/selectable ring sounds include up to 8 melodies and 3 tones can be selected by user. Also support of Distinctive Ringing " 20 distinctive rings can be configured by admin for different types of calls.

H.323 and SIP are supported. G.711 and G.729A codecs are supported.

Documentation/Help

The Administrator Manual gives all the instructions on how to install, configure, and use the phone, as well as maintenance, extended administration features, and using the web-based admin tool. This has a nice 4-page troubleshooting guide which includes general problems, common problems and fault-finding tips.

The Operating instructions include basic operating instructions, help with setting up the phone to a new user's preferences, and detailed use of the features. The troubleshooting information included in this manual involves explaining error messages to the user, a reason the error might occur, and what they can do to help solve it (most are "see the administrator").

Manual Testing

The phones were manually exercised by CT Labs evaluators to verify as many of the product's core features as possible. These tests were designed to take no more than 1.5 hours of tester time. This manual exercise included a two-person live speech quality audit that provided a brief subjective evaluation of the perceived quality. If the IP Phone product supported a hands-free or speakerphone option, then it was also tested and the speech quality was separately assessed by the same live testers.

Manual Registration with these Siemens units is not explained well in the documentation; once it's set up, it works fine. "Join" has to be activated on the phone (default is off) to do 3-party conference calls. Many of the standard features require a server, which is only natural for a PBX maker like Siemens.

One weird thing. If you're on handset you can switch to speaker. Once you do, you can't hang up the handset again without hanging up the call. We also tested an external loudspeaker/microphone from Siemens that will turn a regular optiPoint 600 office phone into a conference phone " very nice!

As for our Speech Quality exercise, two live testers are used to perform this test. Each tester uses one of the IP Phone products supplied by the same vendor. One call is placed between the phones and a conversation is conducted that lasts a minimum of 3 minutes. Due to time limitations on these tests, this is performed using the G.711 codec only under clear network and degraded network conditions.

The test calls were conducted in normal voices and normal levels. The testers attempted to position the handset microphone element a uniform distance from the tester's mouth, typically 1" away such that the microphone element was off to the side of the tester's mouth (to minimize excessive breath noise).

Each tester kept their own test results sheet in front of them, registering the anomalies that they heard as the calls were in progress.

These phones performed extremely well during these tests. Only one tester noted that there was a slightly tinny nature to the sound of the speech during handset and speakerphone modes and a tiny bit of graininess. No significant quality difference was noted between tests performed with clear IP channel and with moderately degraded IP channel conditions.

The speakerphone mode on these phones is effectively half-duplex, which is similar to using a walkie-talkie " that is, if you speak while the other side is also speaking, then the far-side audio gets muted. However, the phones handle the change very quickly, so that even small pauses during near-side speaking provide an opportunity for the far-side audio to break through. Thus, the user experience was closer to normal handset communications.

These phones have very fast "half duplex" audio switching, i.e. just a slight pause in near-side speaking allows far-side audio through.

Automated Speech Quality Tests

CT Labs performed end-to-end phone call tests with two IP phones to assess the overall speech quality level of the product. All tests in this section were performed using automated tests with two of Sage 935AT Communication Sets. These test sets provide two basic suites of tests applicable to this type of product:

1. The "SMOS" test, which uses a special technique to estimate the perceived quality of speech. As defined by Sage, an SMOS number above 4.0 is considered to be toll quality. An SMOS number between 3.0 to 4.0 is considered to be communication quality (intelligible but unnatural, or could be annoying, etc.). An SMOS number below 3.0 is unacceptable for voice communication.

The SMOS test also measures latency. Sage's proprietary SMOS algorithm measures round-trip delay (latency) over a range of 0.0 to 5000.0 msec with an accuracy of 0.2 milliseconds. The measured delay includes delay from a number of sources including voice processing, packetizing the voice into frames, and the jitter buffer process.

2. The "PVIT" test, which measures speech clipping (useful for verifying voice activity detectors), comfort noise, frame slips, and noise floor performance. This test is run for 15 minutes and produces the following measurements:

Frame slip - Voice Packet Jitter/Jitter Buffer Resizing: Jitter occurs when a network jitter buffer dynamically readjusts the buffer size. Buffer resizing balances the conflicting aims for less frame loss and shorter end-to-end transmission delay. A positive (contracting) slip occurs when a voice frame is deleted. A negative (expansive) slip occurs when a filler frame is inserted. PVIT time stamps each voice frame slip, and measures its duration.

Voice Clip-Speech Clipping/Silence Suppression: PVIT provides detailed diagnostic information on key packet network characteristics that impact voice clarity. It reports a voice clip when the signal corruption is at the leading edge of a voice segment. Each voice clip event is measured for duration and time stamped. A running calculation also presents the average duration of all voice clip events.

Noise Hit-Comfort Noise: If, during any individual silence period during the test, the noise level exceeds -45 dBm (45 dBrnC), a Noise Hit event will be reported by PVIT.

The Shunra Storm WAN emulator was used to degrade the network connection between the two IP phones being tested with packet loss, latency, and jitter as indicated in the table below. CT Labs ran the SMOS test using the Sage units for about 5 minutes under each WAN condition. If the Sage units had difficulty synchronizing, an SMOS reading would not be returned for that measurement attempt. Thus, the worse the WAN impairments, the fewer SMOS readings would be returned for a given test period. CT Labs ran each test long enough to get a minimum of 5 valid SMOS readings for each product test condition.

For more on the automated test procedures, including notes on how did it and details on the SMOS and PVIT tests, see .

The following results are presented for the SMOS tests performed with the G.711 CODEC.

The latency performance by the optiPoint 600 phone was found to be very good. An increased jitter buffer for each phone likely explains the extra 48ms of end-to-end latency for the degraded network conditions. The SMOS scores under clear and degraded network conditions were excellent with essentially no difference between clear and degraded network conditions.

The following results are presented for the PVIT tests performed with the G.711 CODEC.

The optiPoint 600 phone recorded 11 frame slips, but this was found to not significantly affect the voice quality score, both in automated and manual testing. There were no noise hits or voice clips detected during the test. Overall this is very good performance, but only a few frame slips are expected for G.711 with no impairments so these PVIT results are merely par for the course.

The following results are presented for the SMOS tests performed with the G.729A CODEC.

The following results are presented for the PVIT tests performed with the G.729 CODEC.

The optiPoint 600 phone recorded zero frame slips but had a moderate number of voice clip and frame loss events during the 15 minute test period. The voice clip events were of a short duration that is typical of G.729A and was not found to significantly degrade the voice quality score. The frame loss measurements recorded by the Sage 935AT could be voice clips or frame slips of greater than 20ms; this is not unusual for a typical G.729A implementation either.

Overall, the optiPoint 600 office SIP phone scored a very good 9.0 for our demanding speech quality performance tests.

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