Session Initiation Protocol Phones

The cost of voice over IP phones is beginning to drop, thanks to widespread adoption of Session Initiation Protocol. We tested three budget-priced SIP phones. Find out which one got

July 22, 2004

11 Min Read
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SIP Phone Test SetupClick to Enlarge

Our interoperability tests featured Pingtel's SIPxchange SIP server, which we thought would be a good reference for SIP implementation because Pingtel has been in the business for some time and recently made its SIP server an open-source product (you can download it at In addition, we used Pingtel's Xpressa SIP phone in our test bed. In general, we found that the phones we tested can handle features like hold, transfer and conferencing without any difficulties, even in a mixed, multivendor environment. However, message-waiting notifications posed some problems. We left voicemail messages for all the phones, and we couldn't get the MWI (message waiting indicator) to light on the ipDialog and Zultys phones. We worked with ipDialog, Zultys and Pingtel but were unable to solve the problem.

These less expensive phones have smaller screens and fewer feature buttons. And though they don't support a wide range of codecs, they do support both the G.711 and G.729 codecs for toll-quality audio. Most of the higher-priced phones we tested last year had full-duplex speakerphones, but only ipDialog offers a full-duplex speakerphone at this low price.

All three phones have as much as or more functionality than you'd find on a low-end business phone. For example, the Snom 200 and the ipDialog SipTone II Ethernet Phone both have LCD screens for Caller ID. On legacy phone systems, this feature typically is found only on more expensive digital sets. However, the phones we tested lack some features you'd expect on an enterprise IP phone. For example, Zultys' ZIP 2 doesn't support the IEEE 802.3af PoE (Power over Ethernet) standard--you must plug the phone into a power outlet, and you can't manage the phone's battery backup from the wiring closet. Still, the ZIP 2 is priced at less than $100.

None of the products provide much diagnostic call-quality information, and none let you verify or set Ethernet duplexity settings. If there is a mismatch between the phone and a desktop switch, or between the phone switch port and the connected desktop PC, you can't verify or change it at the phone. On the plus side, all the phones let you set up a dial plan. This feature is common in legacy PBXs: With a dial plan, the system knows the patterns of typical phone numbers and makes the appropriate connection when the caller finishes dialing. Without this feature, you'd have to hit a "send" key after dialing a number. Finally, all the phones we tested can connect to one another using both the G.711 codec and the bandwidth-saving G.729 codec.

After extensive tests, the Snom 200 emerged as the clear winner, earning the highest marks for interoperability, features and performance.Although the Snom 200 didn't shine in our previous review (it finished fifth out of six products), it fared much better against the phones in this price range. Snom listed the price of the phone at $199, which satisfied our $200 requirement. However, that $199 turned out to be the wholesale price. Distributors list it for $265, $86 more than the ipDialog phone ($179 retail). Yet the Snom phone's features easily justify the higher price tag. For starters, it has a two-line LCD screen that provides dynamic pointers to buttons directly beneath the screen. With these buttons, users can transfer calls and set up conference calls. Zultys' ZIP 2 doesn't have an LCD screen; the SipTone II does, but it is used mostly for Caller ID.

In addition, the Snom 200 has five lighted function keys. You can program speed dials on these keys, and the lights help you keep track of callers during a conference call (a key is lit for every connected party). The Web interface provides easy access to the button programming and other phone features. Although all the vendors offer Web interfaces for their phones, only Snom provides both user and administrative access. This can help you prevent users from changing the phone's configuration settings.

Interoperablity ResultsClick to Enlarge

Also unique to the Snom 200 is its trace feature--a packet trace of all the SIP messages sent to and from the phone, which makes troubleshooting problems as easy as shooting fish in a barrel. You can even cut and paste the trace information to an e-mail if necessary.The Snom 200 scored high on NAT (Network Address Translation) support. Problems with SIP and NAT arise when the IP address is mined from the SIP layer during a call instead of the IP layer. In a NAT environment, the IP address inserted into the SIP layer is taken from a locally assigned address and can be an unroutable private address, in which case the call will not be completed. Snom offers two solutions to this problem. First, it supports UPnP (Universal Plug and Play), which lets an end device communicate with a DSL/cable modem router to derive the true external address, then inserts it in the SIP header. It also supports STUN (Simple Traversal of UDP through NATs), which uses an external server to sort out the addresses. To top it off, the Snom 200 had the lowest delay and the highest quality scores for our baseline test and most of our impairment tests.

Snom 200. Snom Technology, (972) 831-1600. The SipTone II participated in our previous SIP phone tests, and its retail price gained it entry into this review as well. It comes with many of the features you'd expect in an enterprise phone, including support for PoE and an extra switch port for a desktop PC. It has an LCD display showing Caller ID and the phone's operational status. In addition, it has a speakerphone and separate buttons for hold and conference features. The SipTone II was the only phone we tested with a full-duplex speakerphone, and this was enough for it to edge out the ZIP 2 for second place.

However, the SipTone II doesn't support Layer 2 or Layer 3 QoS (quality of service)--the others do. The vendor told us, as it said a year ago, it will add QoS support in a future version. Likewise, the SipTone II fell flat in supporting NTP (Network Time Protocol). NTP automates the clock settings, which logs when calls are placed and received. The SipTone II does offer a call-log feature, though.

The Web interface accesses mixed user and administrative functions, but we wouldn't recommend giving end users access to it. They could too easily misconfigure the phone's settings. The SipTone II supports G.711 and G.729, but its end-to-end delay for both was much higher than the other phones, at 152 milliseconds and 208 milliseconds, respectively.

SipTone II Ethernet Phone. ipDialog, (408) 830-0800. The ZIP 2 phone, at $100, was about half the price of rivals. But for an enterprise installation, it has serious shortcomings. For example, it does not support PoE, and it has only one LAN connection. You'll need a separate cable for a PC. In contrast, both the Snom 200 and the SipTone II have extra PC ports. Zultys' higher-end SIP phones, the ZIP 4x4 and the ZIP 4x5, come with four extra Ethernet ports.We tested the ZIP 4x4 last year (it finished fourth), and we had heard all about the 4x5 in our recent VoIP RFI. Although the sleek 4x5 has built-in encryption capabilities, an analog port for a local 911 connection and Bluetooth capabilities, the lowly ZIP 2 doesn't even have a headset port.

This Zultys phone is better suited for a service provider implementing phones for home users, who would likely plug it into a cable/modem sharing device without PoE but with extra ports to compensate for the lack of a built-in port for the phone. The ZIP 2 does support STUN to deal with NAT.

The ZIP 2 supports both G.711 and G.729, and the vendor says it supports silence suppression for both codecs. Silence suppression cuts down on bandwidth use by filtering out background noise and transmitting packets only when someone is talking. Zultys was the only vendor to also support comfort noise when using silence suppression. Like an aural teddy bear, comfort noise provides background noise for the caller on the other end, assuring him or her that the call is still active. Users can't access the Web interface; it's available only for administrative purposes.

The phone keypad is limited on the ZIP 2 as well. It doesn't have an LCD screen and has only a few feature key buttons that, with the exception of the hold key, are activated by hitting a function key that temporarily changes the function of one of the number keys.

ZIP 2. Zultys Technologies, (408) 328-0450. www.zultys.comPeter Morrissey is a full-time faculty member of Syracuse University's School of Information Studies, and a contributing editor and columnist for Network Computing. Write to him at [email protected].

Delwyn Lee is a graduate student in the Telecommunications and Network Management Program at the School of Information Studies at Syracuse University.

With voice over IP, making a long-distance phone call can be as simple, and inexpensive, as sending a trans-Atlantic e-mail. But the hardware for a VoIP call--a SIP phone--can be cost-prohibitive, unless you use one of the low-end models.

We tested three inexpensive SIP (Session Initiation Protocol) phones: ipDialog's SipTone II, Snom Technology's Snom 200 and Zultys Technologies' ZIP 2. Each one offers Ethernet connectivity. The phones had to connect to each other as well as to our control phone. We compared voice quality, performance, features and price.

The Snom 200 took top honors, with the best performance and attractive features, such as an easy-to-use LCD interface and quick conferencing and call transfers. However, as lower-priced SIP phones go, you still get what you pay for.

packet loss test G.711 CodecClick to Enlarge

We used Pingtel's SIPxchange server 2.4.0, an IP PBX, to provide connectivity among the phones. With Shunra Software's Storm network-emulation software, we created an environment rife with jitter and packet loss. Hewlett-Packard's ProCurve 2650-PWR switch sat at the center of our test network and provided 802.3af PoE to the phones.

To gauge voice quality, we used Agilent Technologies' VQT (Voice Quality Tester) Telegra R and VQT phone adapters. The Telegra R transmits a wave file and compares the quality of the original file to the other phone's received file. It then applies a MOS (Mean Opinion Score) value based on the PSQM algorithm.

jitter test G.711 CodecClick to Enlarge

To test performance, we first conducted a baseline test directly through the HP switch for the G.711 and G.729 codecs with no network impairments. Then we measured the delay and MOS directly through the switch to determine the delay and baseline MOS without impairments (for details on the delay tests, see our charts.)For our interoperability tests, we used the Pingtel SIPxchange for all SIP services and voicemail, and a Pingtel Xpressa phone for our conference-call test. All the phones could make calls to each other using G.711 and G.729 codecs, and both the Snom 200 and SipTone II were interoperable with the 802.3af power on the ProCurve switch.

phone to phone delay

Click to Enlarge

However, the SipTone II and Zip 2 phones could not activate the Message Waiting Indicator for voicemail because they couldn't successfully complete a "subscribe" transaction with the Pingtel server.

--Delwyn Lee


SIP Phones

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