IP PBXs

The two most common explanations vendors gave for features not working were that the third-partyphone does not support that feature or that the SIP standard does not support that feature. There is some truth to these statements. For example, IP PBX vendors had trouble delivering distinctive ring tones to the third-party phones under test. Ring tones are generated locally by the SIP device, so they depend on how the phone manufacturer implements them (for a list of third-party phones used, see "IP Phone Choices," right).

As for SIP, it's young and still growing (see "VoIP: Join the Party" for more on SIP's progress). For example, the SIP standard doesn't detail a signal when a phone goes off hook, so we wanted to see how vendors would handle that. In tests, we placed a call over the Avaya system to a Cisco IP phone we had taken off hook. The phone rang instead of generating a busy signal and forwarding the call to voicemail. However, the same test with the Grandstream and SNOM phones on the Avaya system sent the call to voicemail, which is what we'd expected. An extension of RFC 3265 will address this problem, and Avaya told us it will accommodate the feature in its next release. The other vendors did not get back to us. But note that the presence capabilities of soft phones in this review could identify when their "bound" phone was off hook and generate a SIP "notify" signal for a "busy" status.

None of the systems received a perfect score in our interoperability tests (see "How We Tested SIP-Compliant IP PBXs,"). Although Avaya's came out on top, it had difficulties with managed-call and blind-call transfers on the Cisco phone. Zultys' setup suffered from its choice of Microsoft Messenger as an endpoint and for bring- ing only two third-party SIP phones. Vonexus also didn't bring the required phones to test, and it got off to a false start with an installation problem stemming from insufficient HMP (Host Media Processing) resources to satisfy our test scenarios; in addition, it didn't include the add-on conferencing module specified in our RFP. Finally, Vonexus was the only vendor to hazard testing using an analog phone, an Audiocode eight-port FXS gateway.A word on HMP--this feature, built into Intel CPUs, enables audio processing without the hardware that normally performs this function. A license key must be entered to turn HMP on, which adds one more thing to account for in an installation. If you find that features such as conferencing inexplicably don't work, make sure HMP is on.

We haven't figured out whether the computer will become the next phone or vice versa. But we are willing to bet that the phone will be the next killer application in hardware or software, so we carefully examined each vendor's PC-based applications. We liked Zultys' MXIE best, thanks to its implementation on the Windows desktop and good call- and presence-management features.

We downloaded MXIE directly from Zultys' IP PBX appliance, the MX250. We could localize MXIE for emergency services, such as E911, and configure it with call-handling logic, such as call forwarding and follow-me/find-me rules. It supports 16 simultaneous call appearances and multiple codecs to configure call optimization. In addition, it comes with TAPI (version 2.1) support to hook into Windows applications and installs its own print-to-fax Windows driver. Although MXIE could bind to and easily control call processing for the third-party SIP phones under test, it lacked some UM (unified messaging) features, such as the ability to listen to and delete e-mail messages over the phone, that would have made it a killer.

Avaya's IP Softphone R5 application and Vonexus' Interaction Client also provided a lineup of potential killer applications. Like MXIE, the Avaya Softphone initiates and terminates calls with a click of the mouse. It also implements a secure IM (instant messaging) client with server-side logging and archiving to comply with industry regulations. The Softphone R5 also provides a panoramic view of enterprise users with presence management, just like Vonexus' Interaction Client.

Each vendor advised us of a redundant or failover option for high availability, and each had an answer for the 20 telecommuters accessing the central office from a home-office location. They also made recommendations for basic and executive phones that included user licenses (read the complete vendor responses).

Avaya chose the SNOM 200, priced at about $378 per unit, as both its basic and executive IP phone. Although Avaya says it prefers its own 4602SW SIP Phone (estimated at $284 per unit), we specified a third-party SIP phone to bang on interoperability, and the SNOM 200 fit our minimum requirements of two 100-Mbps ports plus support for IEEE 802.3af and 802.1Q/p. Although a better choice than Zultys' entry, the $200 SNOM 105 (which did not meet minimum requirements and is no longer manufactured), the SNOM 200 lacks a hard key for call transfers, forwards and holds, and was also without a key for volume control. We did find all these features in the Avaya 4602SW SIP Phone and Vonexus' basic phone choice, the Polycom 300 ($215 per unit). All the basic phones supported the G.711 and G.729 audio codecs.

We found these IP PBX systems much easier to manage than their predecessors. Each vendor supplied a Windows 32-bit application to manage all aspects of call processing, users, phones and more. Each came with a database to store all information about users and devices, as well as facilities to view a graphical topology of the system and run standard and configurable reports. However, none of the systems could monitor the health of third-party SIP endpoints, and none supplied us with real-time information on network health or call quality.

Avaya, with its long history in telecommunications, delivered the most mature and feature-rich interface to manage its S8300, the Avaya Site Administrator. Unlike the other vendors, Avaya provided a rich CLI (command-line interface) in addition to a GUI. The combined CLI-GUI was more difficult to master than the GUIs supplied by Zultys and Vonexus, but it gave us an extensive administrative toolset.After examining the vendors' RFPs and performing our hands-on testing, we gave Avaya's SIP Telephony Solution our Editor's Choice award. It provided the best interoperability with third-party phones and the best features-price balance. Zultys came in second; it earned only a fair interoperability score but almost made up the difference by supplying the best soft phone (MXIE) and the best price in our roundup. Vonexus came in just behind Zultys with a strong showing in presence management and UM features. Its product will make management and administration easy for any Windows shop, but it came up short on interoperability and long on price.

Avaya's IP Telephony Solution combines a G700 Media Gateway, an S8300 Media Server and an S8500 Converged Communications Server (CCS). The company's Communications Manager (CM) provides call processing and control from the media server to support analog, digital, H.323 and SIP endpoints through the media gateways and servers. Support for multiple endpoints gave HNF the option to keep some analog handsets until prices for SIP phones drop, later this year, we hope.

The CCS lends SIP support to the media server and gateway, and extends ACD (automatic call distribution), voicemail, IM and UM features to SIP endpoints. It also can serve SIP trunks to a carrier service to take advantage of future SIP-based call origination and termination. The CM supports up to seven-digit dial plans, and Avaya says the system can expand with additional gateways and communication servers, while retaining centralized user and phone administration to reduce operating costs.

Avaya proposed an SG 200 Gateway with VPNremote Client software to connect HNF's home users to the central office securely via two-factor user authentication and 3DES encryption. The SG 200 Gateway supports stateful, multilayer packet inspection and protects against DoS (denial of service) attacks. It also includes a built-in bandwidth manager for VoIP streams and supports encrypted SIP signaling. Like Zultys and Vonexus, Avaya recommended the use of its PC application, Avaya's IP Softphone R5, to supply IM and presence management and facilitate call management.

IM and presence management are enabled by the CCS and the IP Softphone R5 application. As with the other setups, the user license for most phones includes a soft-phone application. We could configure Avaya's soft phone for real-time voice communications using SIP, H.323 or even an analog line. Unlike Zultys or Vonexus, Avaya supports secure IM; the server also provides IM logging and archiving to comply with industry regulations.

Avaya's UM functionality came close to the full-featured product supplied by Vonexus and easily surpassed Zultys' UM implementation. But unlike rivals, Avaya's UM integrates with Exchange and IBM Lotus Domino so users can employ their contact lists to initiate calls and chats. We could create, send and listen to e-mail messages as well as access contacts, calendars and tasks with a TUI (Telephone User Interface) via text-to-speech functionality. And Avaya, like our other participants, includes a "follow-me, find-me" feature based on user-defined rules.

We especially liked Avaya's "Extension to Cellular" option for mobile employees who primarily use cell phones. Users can map their desktop phone and cell phone together as a single extension, which means all incoming calls ring both phones and outgoing calls appear as if they're initiated by the desk set.

In our hands-on portion of the testing, we managed the S8300 Media Server using the Win-32 Avaya Site Administration (ASA) application. ASA provided both a GUI and a CLI to configure the network, call groups, stations and system settings, including date, time, dial plan and class of service. We viewed, ran and scheduled all tasks from a directory tree; popular tasks are also grouped in a favorites bar. We liked the window that displayed a history of our commands with trace information about their execution on the server or on endpoints.

Like the other vendors' management apps, ASA gave us handy wizards for common tasks like adding, changing and removing users and stations. Although ASA let us manage all phones and their services, the interface unfortunately didn't distinguish phone models and their feature sets--we had to make sure we knew which station and feature options were available on our test phones and configure them manually. But this was true for Zultys' and Vonexus' offerings as well--automatic discovery of third-party SIP endpoints and their feature sets was not available from any system tested.We managed the CCS from two separate Web browser pages, one for administering users and resources, one for server maintenance. The SIP Management pages let us import and export users, manage extensions, start and stop services, and configure settings for domain access and services like IM log settings. The Maintenance Web pages deal with hardware configuration, diagnostics and upgrades as well as data backup-and-restore and security. This bifurcated arrangement enabled distributed administration. The CCS also supports a firewall and a Tripwire tool that monitors and audits files on the server to verify they have not been modified or deleted.

We could integrate multiple G700 Media Gateways into single logical service for redundancy. If one failed, traffic was rerouted over surviving links automatically using Spanning Tree or port-redundancy strategies. A duplicate S8300 Media Server ($2,000) also can provide redundant call control. Finally, for $6,200 the CCS may be deployed in a duplicate server configuration where each server shares the same IP address to present a uniform interface to SIP clients when a failure occurs.

Avaya SIP Telephony. Avaya, (800) GO-AVAYA. www.avaya.com

Of the third-party phones Zultys submitted, only the executive model, the Polycom SoundPoint IP 500, met HNF's requirements. Its basic phone, the SNOM 105, does sport two 100-Mbps ports, but it lacks support for IEEE 802.3af and 802.1Q/p. Maybe Zultys thought the SNOM 105's low price--$200 with software license--could make up for these shortcomings.

Zultys' submission for an attendant phone, the SNOM 220, fulfilled only our port and power requirements; the 220 does not support IEEE 802.1Q/p, and in tests we couldn't select an outgoing trunk to send a call nor choose an incoming trunk to answer or redirect a call.On the plus side for Zultys is its MXIE, the best PC-based application we tested. MXIE let us add attendant console functions to any SIP phone, and it can handle as many as 16 simultaneous calls, both internal and external, including calls rerouted from home telecommuters. When we logged in as an operator, the interface listed all users in our test "enterprise" in a graphical display with presence management indicating who is on the phone, out to lunch or free to chat. We transferred calls simply by dragging and dropping call appearances on displayed uses. Alternatively, MXIE provides 10-key access to call-session controls, such as transfer, hold and disconnect, so operators can manage calls without a mouse. And MXIE is not only for operators. Zultys' app, like Avaya's Softphone, is included with the software license for each SIP phone.

MXIE also provided presence management and IM functionality. Presence management and caller status can be automatic or manually driven, based on triggers, such as keyboard or phone activity. We could initiate and terminate calls, set call-handler rules, view the presence of other users, send instant messages, and access voicemail and faxes from our desktop. We could even configure MXIE to simultaneously ring multiple phones.

Zultys' telecommuting solution was clean and simple. Using the ZIP 4x5 phone in home offices, HNF's telecommuters establish IPsec tunnels to the central office's MX250. A VPN concentrator also may be inserted into the equation, but that was not included in Zultys' proposal. At home, the ZIP 4x5 phone provides the functionality of a router with a firewall, DHCP and NAT support. Home users can connect the ZIP 4x5 directly to a cable or DSL modem and set up a home network around it. The phone includes an analog port in case of E911 emergencies and support for a Bluetooth wireless headset and voice-activated dialing. With this design, HNF may set up even more home workers.

The MX250's Linux-based OS performed well. The device uses mirrored SCSI disks, and we could configure it with a standby unit. It also can support an N+1 cluster configuration of as many as four MX250s. The single MX250 system proposed to HNF scales up to 500 users with no additional hardware necessary.MX250 Enterprise Media Exchange, ZIP 4x5 IP Phone. Zultys Technologies, (408) 328-0450. www.zultys.com

Vonexus, like Zultys, built its proposal from the ground up around open standards and SIP. The system is as feature-rich as our Editor's Choice, but it costs more and lagged in our interoperability tests. In addition, Vonexus didn't bring the third-party SIP phones we'd asked for, and the systems lacked HMP support, which means some features, such as conferencing, may not work.

Vonexus was the only submission to use Windows as the host OS in its IP PBX, the EIC (Enterprise Interaction Center). This makes it easy for most enterprises to manage. EIC enables NTLM security, provides an easy path for application integration and uses WMI for event reporting and alerting. But don't forget the care and feeding required to block viruses and malicious code--imagine if not only your data-processing systems but also your voice systems were possible attack vectors.

Vonexus advised HNF to use the Polycom 300 as a basic phone. This was the best basic model in our roundup and had the lowest cost. Unlike Zultys' basic entry, it satisfied our minimum criteria for power, ports and QoS (quality of service). It also had hard keys tied to transfer, forward and hold calls with a dedicated volume-control button. Vonexus, like Zultys, used Polycom's IP 500 phones for executives. Avaya maintained both courses (basic and executive) with the SNOM 200. It appeared Avaya was stacking the deck for its real preference: the Avaya 4602SW.

Like Zultys, Vonexus added its user interface application, dubbed Interactive Client, to its attendant console choice, the Polycom IP 500. Even though the IP 500 has a venerable set of features, IC ratchets it up a few notches. For example, the phone is limited to three line appearances. (Line appearances are simply call appearances. For instance, while talking to one person, another calls. If you place the current call on hold and take the next call, you now have two call appearances on your phone set.) But with IC, line appearances are limited only by the call resources on the server. And, we could preserve server resources by limiting call resources on a per-user basis using a class-of-service configuration. Also, even though the phone does not support any bridged appearances, IC will give it a view of 10 or more bridges.

Vonexus' telecommuter offering was streamlined--the only item required was IC, which Vonexus included in its pricing for the executive but not basic phones. IC costs $45 per user as an add-on to basic phones; factor in that cost for the added functionality. Home users can use their home phones, cell phones or an IP phone with IC. With the EIC's "remote agent" functionality, we could route IC client connections through a secure VPN, and associated phones appeared as a direct extension off the EIC.

Vonexus' presence management let us camp on someone's busy line; in other words, when the line became free, we automatically connected to it without having to redial. In addition, we could update presence using multiple methods, such as changing status via IC or a TUI. Users with the appropriate rights can update the status of others as well. And, unique to this roundup, presence management is integrated with SMS (Simple Messaging System), which shows user status on Web pages. Vonexus says it plans to release a feature pack later this year that will enable integration with groupware and calendaring.

Vonexus gave us two redundancy options. If only basic call routing is required when a system is down, HNF can use a backup proxy server from Vonexus that load balances incoming calls on a per-DID basis and routes them to the proper locations. Maximum redundancy is achieved by deploying an automatic switch-over system in the event of a planned or unplanned outage.

Enterprise Interaction Center. Vonexus, (888) 817-5904. www.vonexus.comSean Doherty is a technology editor and lawyer based at our Syracuse University Real-World Labs®. A former project manager and IT engineer at Syracuse University, he helped develop centrally supported applications and storage systems. Write to him at [email protected].

Peter 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].

We issued an RFP seeking SIP-enabled IP PBXs and required that vendors set up their systems in our Syracuse University Real-World Labs®. We asked them to bring three SIP (Session Initiation Protocol) phones from three third-party vendors as well as an additional third-party endpoint, such as a videoconferencing unit or soft phone. Unfortunately, no one was adventurous enough to try videoconferencing. Avaya and Vonexus brought Xten's X-Pro soft phone, while Zultys Technologies went with Microsoft Messenger 4.7. Note that later versions of Messenger no longer allow the optional Communications Service Account for SIP communication.

In addition to the endpoints, the vendors brought an array of phones from Cisco Systems, Grandstream Networks, Polycom and SNOM Technology AG (see "SIP Phone Choices," left). Avaya played by the rules and used Cisco, Grandstream and Polycom phones. It also showed off its own Avaya Model 4602SW, but since it's not a third-party SIP phone, the 4602SW wasn't used for testing. Vonexus and Zultys did not ante up the right number of phones: Vonexus used Cisco, Polycom and an analog phone with an AudioCodes FXS gateway. Zultys stayed with Polycom and SNOM.

After the vendors set up their systems, we ran a battery of tests to rate the PBXs' ability to interoperate with third-party SIP phones (see "Test Battery,"). We evaluated each phone for call initiation, termination, forwarding, conferencing and more. We also scrutinized presence capabilities and interrogated maintenance features for MAC (moves, adds and changes), user and phone management. After knocking ourselves out in the lab, we compared the RFPs (see our RFP and full vendor responses).

All Network Computing product reviews are conducted by current or former IT professionals in our Real-World Labs® or partner labs, according to our own test criteria. Vendor involvement is limited to assistance in configuration and troubleshooting. Network Computing schedules reviews based solely on our editorial judgment of reader needs, and we conduct tests and publish results without vendor influence.

Reality TV's popularity shows no sign of waning. That may be bad news for America's collective IQ, but it's a bonanza for HaveNoFear Insurance LLC, which indemnifies the sponsors, producers and staffs of these shows against lawsuits by traumatized on-air participants. The company is still growing, but new competitors are coming online constantly. HNF finally has enough money to purchase a SIP-compatible voice over IP system that it hopes will result in a competitive advantage.

Our updated HNF RFP calls for a SIP-capable IP PBX to handle 220 Ethernet-attached phones, 50 telecommuters, 10 analog fax lines, 48 inbound digital trunks with DID (direct inward dialing) support, 35 outbound digital trunks and an ACD (automatic call distribution) system. Finally, HNF wants to be able to use any third-party SIP phone; we asked for recommendations on basic and executive phones, as well as attendant consoles and conferencing phones.

Each vendor had to accommodate 20 telecommuters working full time from home over broadband connections and 30 road warriors who use office phones but also travel extensively. We specified presence management, conferencing and voice-mail capabilities. Finally, we asked about each vendor's plan for application integration and gathered pricing information.Check out our full RFP and complete vendor responses below.

(vital stats)

• MISSION: Indemnifies makers of reality-TV programs

• GOAL: HNF is ready to test SIP-enabled VoIP systems

• TOTAL EMPLOYEES: 180• NUMBER WORKING REMOTELY: 50; 20 full-time telecommuters, 30 road warriors

• Network infrastructure: LAN-based, with Layer 2 and Layer 3 QoS enabled; 100-Mbps connections to desktops, with 802.3af PoE support; and a gigabit backbone connecting the rest of the network

• Requirements: VoIP system must support at least 220 third-party, Ethernet-attached SIP-enabled IP phones, each with two 100-Mbps ports. All phones must support:

• 802.3af and 802.3q/p and either DiffServ- or TOS-based QOS

• Unified messaging, ACD (automatic call distributor) and presence• Telecommuters/road warriors

Complete Vendor Responses

R E V I E W

SIP Compliant IP-PBXs

Sorry,
your browser
is not Java
enabled

Welcome to NETWORK COMPUTING's Interactive Report Card, v2. To launch it, click on the Interactive Report Card ® icon above. The program components take a few moments to load.

Once launched, enter your own product feature weights and click the Recalc button. The Interactive Report Card ® will re-sort (and re-grade!) the products based on the new category weights you entered.

Click here for more information about our Interactive Report Card ®.