VoIPowering Your Office: The Wild iPBX Roundup, Part 2

This time, we compare offerings from Pingtel and Digium itself—the entity that first brought you open-source VoIP.

 By Carla Schroder
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Last week we began our iPBX comparisons with Fonality and AstLinux. Today we're wrapping up with products from Pingtel and Digium.

One important note about PBXtra and trixbox Pro: Both are designed to be your LAN's primary DNS server. If you want your existing DNS server to remain as your primary, you'll have to set up DNS forwarding. (Read about it here.)

SipX and SIPxchange
SipX is the free-of-cost open source SIP-based iPBX from Pingtel. SIPxchange is their commercial edition. They are very similar, with a few differences. The main ones are that SIPxchange includes a call center module, and that it uses the closed-source commercial GIPS media framework instead of open-source codecs. Neither edition supports the multitude of VoIP protocols (IAX, H.323, MGCP, SCCP, UNISTIM, etc.) that Asterisk supports; just SIP.

SIP has become the popular VoIP protocol: Virtually all VoIP networks, endpoints and gateways support SIP. IAX is a native Asterisk protocol that is not nearly as well supported as SIP. H.323 is the grandmommy of VoIP protocols, and is used mainly by big commercial carriers. It is well-supported in most VoIP gear. SIP is similar to H.323 in the way it works, but is less complex and not nearly as heavyweight.

SipX and SIPxchange claim to deliver superior call quality and support greater call volumes by implementing the SIP protocol the way it's designed to be used. SIP separates the signaling stream from the media stream. The signaling stream handles the job of registering and authentication, and then the audio stream can travel by the most efficient route between endpoints, even bypassing your server entirely. Asterisk makes itself the endpoint, so all call traffic must go through it. Thus, it functions as its own bottleneck. (If you want a snazzy VoIP term for this, it is "back-to-back user agent", or B2BUA.) The server does not do any media transcoding, so that's a CPU-intensive chore the server doesn't have to handle. Since transcoding is almost always necessary, the easy way to handle this is to pair your SipX/SIPxchange server with a media gateway like the Cisco 37xx series, or the Audio Codes MP-108. (Visit Pingtel.com to find a list of certified devices.)

SIPxchange has various levels of vendor support and training. It comes in three editions: Enterprise Communications Server, CallManager, and SIPxNano.

Enterprise Communications Server is pretty much the same as SipX, but it includes a call center module. CallManager is a management interface for a distributed VoIP network with multiple servers or locations. SIPxNano, as the name implies, is a wee appliance that puts Enterprise Communications Server (minus the call center module) on a stylishly small server. This is intended for small shops with up to 30 users.

SipX is free of cost and Free software, licensed under the LGPL. It offers only community support, which is pretty good, plus a lot of documentation. It's a free download that includes the operating system, so it's an easy one-CD install. Administration is via Web interface.

Both products auto-provision a number of phones, such as Polycom, snom, LG-Nortel, Cisco, and Grandstream. Auto-provisioning phones is a huge time saver, so always look for this.

SipX also includes a DNS server. See VoIPowering Your Office: Recovering SipX Passwords and DNS Done Right for more information on integrating it into your network.

Asterisk itself
And now we come to the original iPBX and the beginning of the VoIP revolution, Asterisk. Asterisk is currently available in four editions: AsteriskNOW and plain-vanilla Asterisk, both of which are free of cost and open source, Asterisk Business Edition, and the Asterisk Appliance.

Plain old Asterisk is for gnarly geeks who want to customize it to their own specifications, or who want to keep up with the latest releases, or who want to install it on their own choice of operating system. Once you learn your way around Asterisk's vast herds of configuration files you can make changes and updates pretty fast, and it's easier to copy text files than to re-enter gobs of stuff in graphical forms.

AsteriskNOW is a software appliance that includes the operating system, which is the streamlined and customized rPath Linux. rPath is designed for custom appliances such as AsteriskNOW. It has a different set of commands than true Linux, so you'll have to get used to new ways of doing things. It also includes the excellent AsteriskGUI, which is for administering both Asterisk and rPath Linux.

Asterisk Business Edition comes with support, training, and custom programming options. Unlike the free versions, it undergoes considerable quality control testing. It also comes with the AsteriskGUI.

The Asterisk Appliance comes on a sleek small box about the size of a 12-port switch, and is designed to support up to 50 users. This is based on an embedded Linux, uClinux, and it also includes the AsteriskGUI. It supports up to 8 analog ports, if you need them, or just plain old pure VoIP.

Which one?
Hopefully this series has given you enough information to decide which, if any of these, you want to try. There is no such thing as plug-and-play. There is a learning curve no matter which one you pick—not only in telephony but in IP networking and name services, and bandwidth management. The upside is any of these offer you a wealth of features and flexibility that traditional, way-more-expensive PBX systems don't even come close to offering.


This article was originally published on Oct 1, 2007
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