IP-PBXs built on open source software show promise

Test shows four inexpensive systems target ease of use

In this Clear Choice test of four open source-based IP PBX systems, we found products that are well suited for the low end — defined in this testing as supporting as many as 250 concurrent users. By providing simple installation processes, automatic endpoint configuration and straightforward Web-based management interfaces, these products could go a long way in easing IT fears about dealing with open source VoIP products.

ProductSIPxchange ECSSwitchvoxPBXtranet.PBX
VendorPingtelFour Loop TechnologiesFonalityEscaux
Price as tested

$7,200* for software and one-year service contract.

$4,355* for software, performance upgrade and one-year service contract.$4,744* for software, appliance and three-year service contract.$16,316* for software, appliance and one-year service contract.
ProsSingle CD installation; extensive support for autoprovisioning of third-party IP hard phones; group provisioning; load balances and high availability supported.Offers group provisioning and extension templates; switchboard application available for all users.Full turnkey system, hosted maintenance and management service, simple remote user support.Company provides maintenance and management as a service; offers group user configuration; high availability supported.

Limited remote user support.

No autoprovisioning of third-party IP hard phones; no IAX phone support.Limited adds, moves and changes permitted with model; requires Internet connection for management.Requires Internet connection for installation and maintenance.
Link to our IP-PBX Buyer's Guide at www.nwdocfinder.com/1102
The breakdown Pingtel Four Loop Technologies  Fonality Escaux 
Management 20%4454
Features 20%4544
Interoperability 20%5434
Ease of use 20%4453
Architecture 20%5434
Scoring Key: 5: Exceptional; 4: Very good; 3: Average; 2: Below average; 1: Subpar or not available

While this initial test of these products focused on smaller-scale deployment models, several of the systems tested also lend themselves to larger deployments. They incorporate more-advanced features, such as standby systems for failover and user presence capabilities. All provide full Session Initiation Protocol (SIP) support for endpoints as well as trunks. The management interfaces are simple but also provide detail for troubleshooting and bandwidth control necessary in larger environments.

The vendors that accepted our invitation were Escaux, Fonality, Four Loop Technologies and Pingtel. All four products were built on top of Asterisk, the original open source IP PBX (see the assessment of Digium’s Asterisk offering below). The four vendors in our test demonstrated tangible improvements to existing open source IP PBX base code, especially in their efforts to facilitate installation, management and maintenance with GUIs.

The systems were ordered and provisioned as a customer would procure them. We then put them through their paces looking at management, features, interoperability, ease of use and architecture. Because performance metrics such as voice quality are dependent on the endpoint chosen, it was not a key factor in this test.

Pingtel’s SIPxchange earns the Clear Choice award for triumphing over the field in our endpoint interoperability and architecture categories. In the latter category we examined how the product was designed to work. SIPxchange comprises some of the more common practices found in larger, proprietary systems, such as direct paths for the media streams. This limits the burden on the server and allows for better scalability and reliability. Also garnering respectably high scores in our tests were Four Loop and Fonality, but these companies earned their kudos for different reasons. Four Loop’s Switchvox has advanced features — such as a built-in switchboard — that were better than most. Fonality’s PBXtra was the leader of the pack in terms of ease of use, mainly because standard support includes off-site monitoring and management services.

Here is a product-by-product breakdown of our complete test results.

Pingtel SIPxchange

Of Pingtel’s three open source IP PBXs, we tested SIPxchange Enterprise Communication Server, Version 3.7. Pingtel also has the SIPxNano for fewer than 30 users, as well as a free, downloadable version of its sipX product, which is supported by the SIPfoundry open source community.

We installed SIPxchange from a single CD that included the Linux CentOS 4 and the IP PBX software. The installation requires the administrator to enter information about 15 system and network parameters using a basic GUI; the system was up and running in 15 minutes. A second system was brought online to test failover support.

The core infrastructure of the Pingtel system is completely rooted in SIP, and can be installed on multiple Linux kernels or ported to Sun Solaris. No media streams pass through the SIPxchange server. RTP streams are passed directly between endpoints with just call control messages being relayed to the server. This allows for reduced workload requirements on the IP PBX system, which gives Pingtel an architectural advantage over most. The other servers included with the product — such as authentication, registration and presence servers — are software stacks running on the same physical box but are isolated from the IP PBX from a software perspective. So if one of them has issues, it does not bring down the others.

The administrative Web interface allows for quick and easy setup and management. New users can be provisioned and configured separately or as a group. For large numbers of new users, the interface’s support for a CSV format lets you import settings from an Excel spreadsheet. The proper format for the CSV file is included on the “add users” interface.

Because SIPxchange is exclusively SIP-based, adding gateways and other SIP devices is as easy as adding a user to the system. Outbound and inbound call route plans can then be developed from the default settings and utilized.

For high availability the SIPxchange system can be configured for load balancing and redundancy with multiple servers in multiple locations. When we simulated a catastrophic failover, we found that as long as the endpoint being used supported more then one DNS SRV record, it was able to fail over automatically. Because the calls handled by the Pingtel system are randomly load balanced service was not interrupted for some of the endpoints that were registered to the failover call controller.

Interoperability for SIPxchange is extensive. Certified plug-and-play hard phones include all Grandstream, Polycom and Snom phones; Cisco models 7960, 7940, 7912 and 7905; the Linksys ATA-186/88; Hitachi’s WIP 5000 & WIP 3000; Eyebeam’s softphone; and ClearOne Conferencing phone. We tested Polycom and Snom. Registration was quick and straightforward for all phones with excellent voice quality across all endpoints. For analog and T-1/E-1, Pingtel interoperates with AudioCodes, Cisco, Patton Electronic, VegaStream and Mediatrix gateways.

Four Loop Switchvox

Four Loop offers two versions of the Switchvox v2.6 retail open source-based IP PBX, the Switchvox SOHO and the Switchvox SMB, which we put in the rack for this review. The SOHO is designed for a smaller number of users who don’t need a full feature set. Both systems are built upon the Linux 2.6.12 kernel and Asterisk 1.2.

Switchvox more typically is sold and supported by resellers but also can be purchased online. During the purchasing process customers fill out a questionnaire that gives the vendor basic setting and configuration information. The system comes in a 4U server, preconfigured based on the questionnaire or a prior site survey depending on method of purchase. Digium FXS/FXO analog and T-1/E-1 cards are pre-installed prior to shipping.

As with all the other products reviewed, management of the Switchvox server is done via a Web interface. Users can be provisioned separately or by group. A template similar to the one offered by Pingtel can be created for typical settings for certain devices to save on setup and management time. We got the system out of the box, set up and were placing calls in 20 minutes in our lab.

There is administrative access to all of the features such as voice mail recordings, music on hold, agents and call recording. The Switchvox SMB comes with an integrated, full-featured call center and is coupled with an extensive and smartly laid out Initiated Voice Response (IVR) editor. The editor makes short work of creating a complex IVR tree for the auto attendant or call centers. The IVR was impressive and could hold its own against any commercial offering in the market for ease of use.

Each user has Web-based access to an application called Switchboard, which provides the user, depending on how its settings are centrally configured, with an interface similar to commercial softphones with features such as hold, transfer, caller ID and call park. We placed calls with Switchboard and put them on hold, and recorded some. Call details were accurately displayed in real time based on the calls we placed. If the user is designated to be a supervisor, extended privileges are available, such as the ability to view, record and monitor others’ calls, and pick up ringing calls. SwitchBoard was very similar to the other vendors’ softphones in features and functionality except for one important thing — the product is available free of charge.

Switchvox also has a URL editor for integrating CRM-style databases for use in call centers. CRM integration simplifies the life of a call center agent, allowing it to access customer records as part of the call.

Switchvox scored tops in features because of the strength of its free Web client and the straightforward but highly effective IVR editor. Coupling these features with the product’s meet-me conferencing, recording and monitoring abilities, we felt the systems contained nearly everything a small but sophisticated organization would need in a PBX. Additionally, all features were easy to access, enable and configure via the Web management interface.

In terms of endpoint interoperability, we were able to get all of the phones in our test bed to register with this system and were able to place calls with the same results as the other vendors in terms of voice quality.

Switchvox has no support for IAX endpoints, even though the product is based on Asterisk. Switchvox noted it had never been asked by customers to provide that support.

Fonality PBXtra

Fonality’s PBXtra 3.5 Professional edition appliance is based on CentOS 4.3 and Asterisk 1.2. Fonality sells two other retail versions — Standard and Call Center — and offers a free downloadable open source IP PBX version called Trixbox.

PBXtra is offered as a complete turnkey IP PBX appliance. Customers fill out a simple questionnaire online and the device is preconfigured, provisioned and shipped to the customer site. It is designed for any level of technical knowledge or capability. The only requirements are an internal network and an Internet connection.

Fonality shipped the system to us as it would to a customer. We were able to get it up and running within five minutes. We did have to call technical support for one minor setting for outbound dialing that took all of five minutes to find. It’s notable that two hours of setup support are included with purchase.

An annual software maintenance and support agreement can be purchased and is based on the number of users. The most current user database and configuration information is saved by the hosted site. If connectivity to the Internet is lost, the system will function normally based on its last known settings until the connection is restored.

The vendor externally monitors the health of the IP PBX at all times, and its technical support staff addresses any problems it detects per the service contract agreement. This service certainly takes the headaches out of day-to-day management and maintaining VoIP network uptime.

The PBXtra offers all of the features inherent to Asterisk, as well as some extended, proprietary features, such as the Heads Up Display (HUD), an optional $998 client application that extends call control features to a user’s desktop and is available to all users in the system after a one-time purchase. The HUD feature also provides real-time presence awareness, one-click Outlook integration, barge-in capabilities, record, chat, hold and park features.

HUD privileges are controlled through class-of-service settings controlled from the Web-Admin Interface. Adding extensions and setting permissions is simple to do. A fully functional auto attendant editor is included should an administrator want to customize the system. We exercised the HUD remotely and found it added a rich call center-like functionality for users.

Fonality provisions only approved hard phones from Polycom, Cisco and Aastra as well as PBXtra EyeBeam softphones. It will provision other models for an additional fee, but for the purpose of meeting quality assurance and performance standards, you must send the device to Fonality for setup.

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