Session Initiation Protocol Phones update from July 2004

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 26, 2004

4 Min Read
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Mobile phone users know acceptable service involves three factors: an inexpensive plan, decent coverage and an affordable phone that can handle your needs with panache. Same with voice over IP. VoIP's protocols are coming along, and its coverage area is as big as the Internet, but the high price of VoIP phones has been a deterrent. Now that last obstacle is beginning to disappear. More vendors are adopting SIP (Session Initiation Protocol) as the standard for IP phone connectivity, which is increasing competition and bringing about potentially higher production volumes and lower prices.

With all that in mind, we decided to test low-priced SIP phones. We invited Alcatel, Avaya, Cisco Systems, Grandstream Networks, ipDialog, Mitel Networks, Nortel Networks, Pingtel, Polycom, Siemens, Snom Technology and Zultys Technologies to send their SIP phones to our Syracuse University Real-World Labs. We asked for phones that cost no more than $200. And we made it clear we would test interoperability among all the phones and performance over a simulated network, with degradation. Knowing these guidelines, only three vendors agreed to participate: ipDialog, Snom and Zultys.

The last time we tested SIP phones, we did not restrict participation based on price, nor did we undertake any performance tests. This time, we did conduct performance tests. We used an Agilent Telegra VQT with phone adapters to test voice quality and Shunra Software's Storm network-emulation device to introduce jitter and packet loss into the network. The phones fared well in spite of some poor network conditions. On most Internet connections, all would perform satisfactorily.

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 www.sipfoundry.org). 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.

(Click on Executive Summary of SIP Test below to get more information.)

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